Re: [asterisk-users] Paging systems?

[Evan P. Hall]

If you read the manual carefully, it is talking about connecting to a loop 
start trunk port (FXO), not a station port (FXS).  The paging unit is not 
expecting ring voltage and may get damaged if you send it.  In all of the 
scenarios it outlines, the paging system is generating the battery and dial 
tone and expecting the phone or the PBX to simply go off hook and then dial a 
zone.

You’ll need an ATA with an FXO port (like a Grandstream HT813) and then dial 
the desired zone number out that trunk.

-Evan

From: asterisk-users  on behalf of 
Bryant Zimmerman 
Reply-To: "brya...@granddial.com" , Asterisk Users 
Mailing List - Non-Commercial Discussion 
Date: Thursday, March 21, 2019 at 8:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Paging systems?

Michael

Based on how I read the manual if you connect the TIP and RING of the ATA to 
the right pairs you should be able to send a call to the paging box. It looks 
that when the page call is picked up by the paging system you would then press 
a zone 1-9 or 0 for all.  The page would then bridge to the desired zone. the 
page would complete when the call is hung up. You would likely need to make 
sure the ATA is using current loop disconnect or reverse to ensure hang-up.

I think it should be the PABX config using the Figure 3 configuration.

Best of luck

Bryant Zimmerman

Sr. Systems Architect
Grand Dial Communications, A ZK Tech Inc. Company
616-299-5607 (mobile)
616-855-1030 Ext. 2003 (office)


From: Michael Munger 
Sent: 3/21/19 7:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
, John Novack 
Subject: Re: [asterisk-users] Paging systems?
Excellent point.
This is it: https://www.valcom.com/pdf/v-1109rthf.pdf
Get Outlook for Android<https://aka.ms/ghei36>




On Thu, Mar 21, 2019 at 7:22 PM -0400, "John Novack"  
mailto:jnov...@comcast.net>> wrote:



Michael Munger wrote:
Does anyone have an (overhead) paging system that they like that works with SIP?

We’ve got a client with an old paging system that (supposedly) just takes an 
rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn’t 
auto-answer the call, so paging never happens.

Does it expect to see a POTS line with battery on it?
Then a Cisco or other ATA that would work to supply service to a POTS phone 
should work
OR:
Does it expect to see a POTS connection from a PBX trunk, and supply battery TO 
the trunk?
Then you would need a Cisco or other ATA with an FXO connection.

Both types of paging systems have been made and both styles of connections have 
existed through the last 30 + years, and since you haven't revealed the brand 
and model of paging system it makes troubleshooting difficult.
Using the existing system can be made to work
I use a very old Harris PagePak VS that was used with a Western Electric 
Horizon system back in the dark ages with Asterisk

John Novack



--

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Re: [asterisk-users] Paging systems?

[Bryant Zimmerman]

Michael

Based on how I read the manual if you connect the TIP and RING of the ATA to 
the right pairs you should be able to send a call to the paging box. It looks 
that when the page call is picked up by the paging system you would then press 
a zone 1-9 or 0 for all.  The page would then bridge to the desired zone. the 
page would complete when the call is hung up. You would likely need to make 
sure the ATA is using current loop disconnect or reverse to ensure hang-up.

I think it should be the PABX config using the Figure 3 configuration.

Best of luck

Bryant Zimmerman

Sr. Systems Architect
Grand Dial Communications, A ZK Tech Inc. Company
616-299-5607 (mobile)
616-855-1030 Ext. 2003 (office)


From: Michael Munger 
Sent: 3/21/19 7:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
, John Novack 
Subject: Re: [asterisk-users] Paging systems?
Excellent point.

This is it: https://www.valcom.com/pdf/v-1109rthf.pdf

Get Outlook for Android

On Thu, Mar 21, 2019 at 7:22 PM -0400, "John Novack"   
wrote:

Michael Munger wrote:

Does anyone have an (overhead) paging system that they like that works with SIP?



We’ve got a client with an old paging system that (supposedly) just takes an 
rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn’t 
auto-answer the call, so paging never happens.


Does it expect to see a POTS line with battery on it?
Then a Cisco or other ATA that would work to supply service to a POTS phone 
should work
OR:
Does it expect to see a POTS connection from a PBX trunk, and supply battery TO 
the trunk?
Then you would need a Cisco or other ATA with an FXO connection.

Both types of paging systems have been made and both styles of connections have 
existed through the last 30 + years, and since you haven't revealed the brand 
and model of paging system it makes troubleshooting difficult.
Using the existing system can be made to work
I use a very old Harris PagePak VS that was used with a Western Electric 
Horizon system back in the dark ages with Asterisk

John Novack

--  Dog is my Co-Pilot


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Re: [asterisk-users] Paging systems?

[John Novack]

All you need then is a loop start trunk port. Any SIP ATA designed to connect 
to an ANALOG PSTN line.

The doc you sent doesn't show any rj11, so someone probably installed one to 
the first pair out of the 25 pair cable, blue/white

If you were to plug any old phone into that you should at least hear a live 
connection and dial tone from the box

You will need to send dtmf tones to the box for specific zones, or all call. That may be 
one reason you think but don't get an "auto answer" IF you are using an FXO ATA 
then it is sitting there waiting for some DTMF digits 1-9 for the zones, and zero for all 
call

This is also a "talk back" system. Does the customer expect to use the talk 
back feature?

You will need to put that into your dialplan

All easy to do in Asterisk

Similar but quite different than what I have. I announce incoming call numbers 
through the system, as well as quarterly time announcements through call files 
and a cron job. I use an FXO port off a channel bank, but an ATA with an FXO 
port should do the trick


Enjoy

John Novack

Michael Munger wrote:

Excellent point.

This is it: https://www.valcom.com/pdf/v-1109rthf.pdf

Get Outlook for Android 




On Thu, Mar 21, 2019 at 7:22 PM -0400, "John Novack" mailto:jnov...@comcast.net>> wrote:



Michael Munger wrote:


Does anyone have an (overhead) paging system that they like that works with 
SIP?

We’ve got a client with an old paging system that (supposedly) just takes 
an rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn’t 
auto-answer the call, so paging never happens.



Does it expect to see a POTS line with battery on it?
Then a Cisco or other ATA that would work to supply service to a POTS phone 
should work
OR:
Does it expect to see a POTS connection from a PBX trunk, and supply 
battery TO the trunk?
Then you would need a Cisco or other ATA with an FXO connection.

Both types of paging systems have been made and both styles of connections 
have existed through the last 30 + years, and since you haven't revealed the 
brand and model of paging system it makes troubleshooting difficult.
Using the existing system can be made to work
I use a very old Harris PagePak VS that was used with a Western Electric 
Horizon system back in the dark ages with Asterisk

John Novack

-- 
Dog is my Co-Pilot






--
Dog is my Co-Pilot

-- 
_
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Re: [asterisk-users] Paging systems?

[Michael Munger]

Excellent point.

This is it: https://www.valcom.com/pdf/v-1109rthf.pdf

Get Outlook for Android




On Thu, Mar 21, 2019 at 7:22 PM -0400, "John Novack" 
mailto:jnov...@comcast.net>> wrote:



Michael Munger wrote:
Does anyone have an (overhead) paging system that they like that works with SIP?

We’ve got a client with an old paging system that (supposedly) just takes an 
rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn’t 
auto-answer the call, so paging never happens.

Does it expect to see a POTS line with battery on it?
Then a Cisco or other ATA that would work to supply service to a POTS phone 
should work
OR:
Does it expect to see a POTS connection from a PBX trunk, and supply battery TO 
the trunk?
Then you would need a Cisco or other ATA with an FXO connection.

Both types of paging systems have been made and both styles of connections have 
existed through the last 30 + years, and since you haven't revealed the brand 
and model of paging system it makes troubleshooting difficult.
Using the existing system can be made to work
I use a very old Harris PagePak VS that was used with a Western Electric 
Horizon system back in the dark ages with Asterisk

John Novack


--
Dog is my Co-Pilot
-- 
_
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Check out the new Asterisk community forum at: https://community.asterisk.org/

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] Paging systems?

[Darryl Moore]

No. The SNOM PA1 are headless SIP clients which you configure in auto
answer and connect to an amplifier to drive PA speakers. The phones are
where you make the announcements from.

On Thu, Mar 21, 2019, 5:07 PM Antony Stone, <
antony.st...@asterisk.open.source.it> wrote:

> On Thursday 21 March 2019 at 21:59:51, Darryl Moore wrote:
>
> > For a paging system? No you don't. A number of SNOM PA1's and a few
> > grandstream phones and you're golden.
>
> Are you suggesting using standard telephones (presumably in auto-answer
> speakerphone mode) as paging devices?
>
> Depending on the environment, it can work very well (quiet office) or not
> at all
> (noisy workshop, large factory floor).
>
> > If you do need FXO or FXS, they are just as easy to setup as well, and
> there
> > are lots to choose from.
> >
> > On Thu, Mar 21, 2019, 4:45 PM Ryan, Travis, 
> wrote:
> > > You need more than an ATA. You need something with an FSO and FXO. I’ve
> > > used Linksys/SPA3102-3.3.6 and been happy with it.
> > >
> > > *From:* asterisk-users  *On
> > > Behalf Of *Sebastian Nielsen
> > > *Sent:* Thursday, March 21, 2019 3:01 PM
> > > *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' <
> > > asterisk-users@lists.digium.com>
> > > *Subject:* Re: [asterisk-users] Paging systems?
> > >
> > > How did the page system answer the call when it was used with the
> analog
> > > system?
> > >
> > > You could propably ”fake” those signals from inside asterisk, and cause
> > > it to answer.
> > >
> > > *Från:* asterisk-users  *För
> > > *Michael Munger
> > > *Skickat:* den 21 mars 2019 20:00
> > > *Till:* asterisk-users@lists.digium.com
> > > *Ämne:* [asterisk-users] Paging systems?
> > >
> > > Does anyone have an (overhead) paging system that they like that works
> > > with SIP?
> > >
> > > We’ve got a client with an old paging system that (supposedly) just
> takes
> > > an rj11 POTS connection, but when we put an SPA Cisco adapter on it, it
> > > doesn’t auto-answer the call, so paging never happens.
>
> I'm still intrigued to know how this really was plugged in and how it
> operated.
>
>
> Antony.
>
> --
> All generalisations are inaccurate.
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Paging systems?

[John Novack]

Michael Munger wrote:


Does anyone have an (overhead) paging system that they like that works with SIP?

We’ve got a client with an old paging system that (supposedly) just takes an 
rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn’t 
auto-answer the call, so paging never happens.



Does it expect to see a POTS line with battery on it?
Then a Cisco or other ATA that would work to supply service to a POTS phone 
should work
OR:
Does it expect to see a POTS connection from a PBX trunk, and supply battery TO 
the trunk?
Then you would need a Cisco or other ATA with an FXO connection.

Both types of paging systems have been made and both styles of connections have 
existed through the last 30 + years, and since you haven't revealed the brand 
and model of paging system it makes troubleshooting difficult.
Using the existing system can be made to work
I use a very old Harris PagePak VS that was used with a Western Electric 
Horizon system back in the dark ages with Asterisk

John Novack

--
Dog is my Co-Pilot

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] Paging systems?

[Bryant Zimmerman]

Michael

We use Bogan UTI1 box in conjunction with an ATA to patch to any overhead 
paging system. You patch the box directly into the amp line in for the 
overhead. When you call the extension it answers and puts the audio on the line 
to the PA.

If you only have a few speakers the UTI1 can even handle being the amp for a 
few speakers.

Bryant Zimmerman

Sr. Systems Architect
Grand Dial Communications, A ZK Tech Inc. Company
616-299-5607 (mobile)
616-855-1030 Ext. 2003 (office)


From: Darryl Moore 
Sent: 3/21/19 4:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Paging systems?
For a paging system? No you don't. A number of SNOM PA1's and a few grandstream 
phones and you're golden. If you do need FXO or FXS, they are just as easy to 
setup as well, and there are lots to choose from.

On Thu, Mar 21, 2019, 4:45 PM Ryan, Travis,  wrote:

You need more than an ATA. You need something with an FSO and FXO. I’ve used 
Linksys/SPA3102-3.3.6 and been happy with it.







From: asterisk-users  On Behalf Of 
Sebastian Nielsen
Sent: Thursday, March 21, 2019 3:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 

Subject: Re: [asterisk-users] Paging systems?



How did the page system answer the call when it was used with the analog system?

You could propably ”fake” those signals from inside asterisk, and cause it to 
answer.



Från: asterisk-users  För Michael 
Munger
Skickat: den 21 mars 2019 20:00
Till: asterisk-users@lists.digium.com
Ämne: [asterisk-users] Paging systems?



Does anyone have an (overhead) paging system that they like that works with SIP?



We’ve got a client with an old paging system that (supposedly) just takes an 
rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn’t 
auto-answer the call, so paging never happens.





Michael J. Munger, dCAP, MCPS, MCNPS, MBSS

Microsoft Certified Professional

Microsoft Certified Small Business Specialist

Digium Certified Asterisk Professional

High Powered Help, Inc.

p:

678-905-8569

w:

hph.io  e: m...@hph.io





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Re: [asterisk-users] Paging systems?

[Sebastian Nielsen]

The thing is:

Does the paging system connect to a line (like it was a deskphone) or does
the paging system ACT as a line (that you connect a deskphone to)?

 

If the page system is technically a phone, the it should work with a SPA.
What you need to do then, is to figure out what the line does to get the
paging system to auto-answer.

The first thing you could do, is to connect a regular home-phone to the same
jack that the paging system were PREVIOUSLY connected, and then try ”paging”
it.

 

Then check the display. It could display a specific caller ID (that you need
to fake inside Asterisk) or it could send specific signals (which you hear
on the rings).

If you then send this ”fake” callerid from the asterisk to the SPA, it will
also send out this ”fake” callerid out to the paging system and cause it to
answer.

 

 

Or it could be the opposite, the paging system IS the line, and you
technically connect a line-out port to the paging system, ergo, the page
system acts like a phone company’s line in the wall.

Then you need something with a FXS port (something that acts like a phone).

 

 

 

Från: asterisk-users  För Michael
Munger
Skickat: den 21 mars 2019 21:05
Till: asterisk-users@lists.digium.com
Ämne: Re: [asterisk-users] Paging systems?

 

It worked on the old system.

I am open to suggestions, but don't want (or have the option) to add a TDM
card.

 





Michael Munger, dCAP, MCPS, MCNPS, MBSS


Microsoft Certified Professional


Microsoft Certified Small Business Specialist


Digium Certified Asterisk Professional


High Powered Help, Inc.


p:

678-905-8569


w:

 <https://hph.io> hph.io  e:  <mailto:m...@hph.io> m...@hph.io






On 3/21/19 3:01 PM, Sebastian Nielsen wrote:

How did the page system answer the call when it was used with the analog
system?

You could propably ”fake” those signals from inside asterisk, and cause it
to answer.

 

Från: asterisk-users  <mailto:asterisk-users-boun...@lists.digium.com>
 För Michael Munger
Skickat: den 21 mars 2019 20:00
Till: asterisk-users@lists.digium.com
<mailto:asterisk-users@lists.digium.com> 
Ämne: [asterisk-users] Paging systems?

 

Does anyone have an (overhead) paging system that they like that works with
SIP?

 

We’ve got a client with an old paging system that (supposedly) just takes an
rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn’t
auto-answer the call, so paging never happens.

 

 





Michael J. Munger, dCAP, MCPS, MCNPS, MBSS


Microsoft Certified Professional


Microsoft Certified Small Business Specialist


Digium Certified Asterisk Professional


High Powered Help, Inc.


p:

678-905-8569


w:

 <https://hph.io> hph.io  e:  <mailto:m...@hph.io> m...@hph.io

 

 







smime.p7s
Description: S/MIME Cryptographic Signature
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Re: [asterisk-users] Paging systems?

[Antony Stone]

On Thursday 21 March 2019 at 21:59:51, Darryl Moore wrote:

> For a paging system? No you don't. A number of SNOM PA1's and a few
> grandstream phones and you're golden.

Are you suggesting using standard telephones (presumably in auto-answer 
speakerphone mode) as paging devices?

Depending on the environment, it can work very well (quiet office) or not at 
all 
(noisy workshop, large factory floor).

> If you do need FXO or FXS, they are just as easy to setup as well, and there
> are lots to choose from.
> 
> On Thu, Mar 21, 2019, 4:45 PM Ryan, Travis,  wrote:
> > You need more than an ATA. You need something with an FSO and FXO. I’ve
> > used Linksys/SPA3102-3.3.6 and been happy with it.
> > 
> > *From:* asterisk-users  *On
> > Behalf Of *Sebastian Nielsen
> > *Sent:* Thursday, March 21, 2019 3:01 PM
> > *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' <
> > asterisk-users@lists.digium.com>
> > *Subject:* Re: [asterisk-users] Paging systems?
> > 
> > How did the page system answer the call when it was used with the analog
> > system?
> > 
> > You could propably ”fake” those signals from inside asterisk, and cause
> > it to answer.
> > 
> > *Från:* asterisk-users  *För
> > *Michael Munger
> > *Skickat:* den 21 mars 2019 20:00
> > *Till:* asterisk-users@lists.digium.com
> > *Ämne:* [asterisk-users] Paging systems?
> > 
> > Does anyone have an (overhead) paging system that they like that works
> > with SIP?
> > 
> > We’ve got a client with an old paging system that (supposedly) just takes
> > an rj11 POTS connection, but when we put an SPA Cisco adapter on it, it
> > doesn’t auto-answer the call, so paging never happens.

I'm still intrigued to know how this really was plugged in and how it 
operated.


Antony.

-- 
All generalisations are inaccurate.

   Please reply to the list;
 please *don't* CC me.

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Re: [asterisk-users] Paging systems?

[Darryl Moore]

For a paging system? No you don't. A number of SNOM PA1's and a few
grandstream phones and you're golden. If you do need FXO or FXS, they are
just as easy to setup as well, and there are lots to choose from.

On Thu, Mar 21, 2019, 4:45 PM Ryan, Travis,  wrote:

> You need more than an ATA. You need something with an FSO and FXO. I’ve
> used Linksys/SPA3102-3.3.6 and been happy with it.
>
>
>
>
>
>
>
> *From:* asterisk-users  *On
> Behalf Of *Sebastian Nielsen
> *Sent:* Thursday, March 21, 2019 3:01 PM
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' <
> asterisk-users@lists.digium.com>
> *Subject:* Re: [asterisk-users] Paging systems?
>
>
>
> How did the page system answer the call when it was used with the analog
> system?
>
> You could propably ”fake” those signals from inside asterisk, and cause it
> to answer.
>
>
>
> *Från:* asterisk-users  *För *Michael
> Munger
> *Skickat:* den 21 mars 2019 20:00
> *Till:* asterisk-users@lists.digium.com
> *Ämne:* [asterisk-users] Paging systems?
>
>
>
> Does anyone have an (overhead) paging system that they like that works
> with SIP?
>
>
>
> We’ve got a client with an old paging system that (supposedly) just takes
> an rj11 POTS connection, but when we put an SPA Cisco adapter on it, it
> doesn’t auto-answer the call, so paging never happens.
>
>
>
>
>
> Michael J. Munger, dCAP, MCPS, MCNPS, MBSS
>
> *Microsoft Certified Professional*
>
> *Microsoft Certified Small Business Specialist*
>
> *Digium Certified Asterisk Professional*
>
> *High Powered Help, Inc.*
>
> p:
>
> 678-905-8569
>
> w:
>
> hph.io  e: m...@hph.io
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Paging systems?

[Ryan, Travis]

You need more than an ATA. You need something with an FSO and FXO. I've used 
Linksys/SPA3102-3.3.6 and been happy with it.



From: asterisk-users  On Behalf Of 
Sebastian Nielsen
Sent: Thursday, March 21, 2019 3:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 

Subject: Re: [asterisk-users] Paging systems?

How did the page system answer the call when it was used with the analog system?
You could propably "fake" those signals from inside asterisk, and cause it to 
answer.

Från: asterisk-users 
mailto:asterisk-users-boun...@lists.digium.com>>
 För Michael Munger
Skickat: den 21 mars 2019 20:00
Till: asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>
Ämne: [asterisk-users] Paging systems?

Does anyone have an (overhead) paging system that they like that works with SIP?

We've got a client with an old paging system that (supposedly) just takes an 
rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn't 
auto-answer the call, so paging never happens.


[cid:image001.png@01D4E005.74B786F0]

Michael J. Munger, dCAP, MCPS, MCNPS, MBSS

Microsoft Certified Professional

Microsoft Certified Small Business Specialist

Digium Certified Asterisk Professional

High Powered Help, Inc.

p:

678-905-8569

w:

hph.io<https://hph.io>  e: m...@hph.io<mailto:m...@hph.io>




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Re: [asterisk-users] Paging systems?

[Darryl Moore]

I've used SNOM PA1 before with good success.

On Thu, Mar 21, 2019, 2:59 PM Michael Munger,  wrote:

> Does anyone have an (overhead) paging system that they like that works
> with SIP?
>
>
>
> We’ve got a client with an old paging system that (supposedly) just takes
> an rj11 POTS connection, but when we put an SPA Cisco adapter on it, it
> doesn’t auto-answer the call, so paging never happens.
>
>
>
>
>
> Michael J. Munger, dCAP, MCPS, MCNPS, MBSS
>
> *Microsoft Certified Professional*
>
> *Microsoft Certified Small Business Specialist*
>
> *Digium Certified Asterisk Professional*
>
> *High Powered Help, Inc.*
>
> p:
>
> 678-905-8569
>
> w:
>
> hph.io  e: m...@hph.io
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Paging systems?

[Eric Wieling]

These seem to work well: 
http://www.vikingelectronics.com/product_docs/product_manuals/VE__PA-15__15_Watt_Paging_Amplifier_with_Background_Music_and_Loud_Ringing_486.pdf



On 03/21/2019 02:53 PM, Michael Munger wrote:
Does anyone have an (overhead) paging system that they like that works 
with SIP?


We’ve got a client with an old paging system that (supposedly) just 
takes an rj11 POTS connection, but when we put an SPA Cisco adapter on 
it, it doesn’t auto-answer the call, so paging never happens.




Michael J. Munger, dCAP, MCPS, MCNPS, MBSS

*Microsoft Certified Professional*

*Microsoft Certified Small Business Specialist*

*Digium Certified Asterisk Professional*

*High Powered Help, Inc.*

p:



678-905-8569

w:



hph.io e: m...@hph.io 





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Re: [asterisk-users] Paging systems?

[Nabeel Jafferali]

I've had good experience with Algo's VoIP paging solutions.

--
Nabeel Jafferali


On Thu, Mar 21, 2019 at 3:00 PM Michael Munger  wrote:

> Does anyone have an (overhead) paging system that they like that works
> with SIP?
>
>
>
> We’ve got a client with an old paging system that (supposedly) just takes
> an rj11 POTS connection, but when we put an SPA Cisco adapter on it, it
> doesn’t auto-answer the call, so paging never happens.
>
>
>
>
>
> Michael J. Munger, dCAP, MCPS, MCNPS, MBSS
>
> *Microsoft Certified Professional*
>
> *Microsoft Certified Small Business Specialist*
>
> *Digium Certified Asterisk Professional*
>
> *High Powered Help, Inc.*
>
> p:
>
> 678-905-8569
>
> w:
>
> hph.io  e: m...@hph.io
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Paging systems?

[Doug Lytle]

>>> Does anyone have an (overhead) paging system that they like that works with 
>>> SIP?

Our old phone system back ends into a Bogen AMP.

I'm in the process of replacing that system (Meridian) with Asterisk and found 
that the snom PA1 works very well.  If an AMP is involved, this might work.

http://wiki.snom.com/File:Snom_pa1.png

Doug
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Re: [asterisk-users] Paging systems?

[Michael Munger]

It worked on the old system.

I am open to suggestions, but don't want (or have the option) to add a
TDM card.



Michael Munger, dCAP, MCPS, MCNPS, MBSS
*Microsoft Certified Professional*
*Microsoft Certified Small Business Specialist*
*Digium Certified Asterisk Professional*
*High Powered Help, Inc.*
p:  678-905-8569
w:  hph.io <https://hph.io>  e: m...@hph.io <mailto:m...@hph.io>



On 3/21/19 3:01 PM, Sebastian Nielsen wrote:
>
> How did the page system answer the call when it was used with the
> analog system?
>
> You could propably ”fake” those signals from inside asterisk, and
> cause it to answer.
>
>  
>
> *Från:* asterisk-users  *För
> *Michael Munger
> *Skickat:* den 21 mars 2019 20:00
> *Till:* asterisk-users@lists.digium.com
> *Ämne:* [asterisk-users] Paging systems?
>
>  
>
> Does anyone have an (overhead) paging system that they like that works
> with SIP?
>
>  
>
> We’ve got a client with an old paging system that (supposedly) just
> takes an rj11 POTS connection, but when we put an SPA Cisco adapter on
> it, it doesn’t auto-answer the call, so paging never happens.
>
>  
>
>  
>
>   
>
> Michael J. Munger, dCAP, MCPS, MCNPS, MBSS
>
> *Microsoft Certified Professional*
>
> *Microsoft Certified Small Business Specialist*
>
> *Digium Certified Asterisk Professional*
>
> *High Powered Help, Inc.*
>
> p:
>
>   
>
> 678-905-8569
>
> w:
>
>   
>
> hph.io <https://hph.io>  e: m...@hph.io <mailto:m...@hph.io>
>
>  
>
>  
>
>


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Re: [asterisk-users] Paging systems?

[Sebastian Nielsen]

How did the page system answer the call when it was used with the analog
system?

You could propably ”fake” those signals from inside asterisk, and cause it
to answer.

 

Från: asterisk-users  För Michael
Munger
Skickat: den 21 mars 2019 20:00
Till: asterisk-users@lists.digium.com
Ämne: [asterisk-users] Paging systems?

 

Does anyone have an (overhead) paging system that they like that works with
SIP?

 

We’ve got a client with an old paging system that (supposedly) just takes an
rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn’t
auto-answer the call, so paging never happens.

 

 





Michael J. Munger, dCAP, MCPS, MCNPS, MBSS


Microsoft Certified Professional


Microsoft Certified Small Business Specialist


Digium Certified Asterisk Professional


High Powered Help, Inc.


p:

678-905-8569


w:

 <https://hph.io> hph.io  e:  <mailto:m...@hph.io> m...@hph.io

 

 



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[asterisk-users] Paging systems?

[Michael Munger]

Does anyone have an (overhead) paging system that they like that works with SIP?

We've got a client with an old paging system that (supposedly) just takes an 
rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn't 
auto-answer the call, so paging never happens.


[cid:image001.png@01D4DFF6.9C1F1AA0]

Michael J. Munger, dCAP, MCPS, MCNPS, MBSS

Microsoft Certified Professional

Microsoft Certified Small Business Specialist

Digium Certified Asterisk Professional

High Powered Help, Inc.

p:

678-905-8569

w:

hph.io  e: m...@hph.io




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Re: [asterisk-users] Paging in waves.

[John Kiniston]

I wanted to follow up that I had solved my issue to my satisfaction and
share the dialplan that did so.

[sub-masspage]
exten => s,1,NoOP
same  =>   n,Answer
same  =>   n,Set(filename=PAGE)
same  =>   n,Wait(1)
same  =>   n,Record(pagequeue/${filename}%d.gsm,0,60,yk)
same  =>
n,Set(DURATION=$[CEIL(${STAT(s,/var/lib/asterisk/sounds/${RECORDED_FILE}.gsm)}
/ 1650)])
same  =>   n,Playback(one-moment-please)
same  =>   n,SIPAddHeader(Alert-Info: RingAnswer)
same  =>   n,SipAddHeader(Call-Info: \;answer-after=0); Yealink
same  =>   n,ExecIf($[${ISNULL(${ARG1})}=0]?Dial(LOCAL/GO$[${DURATION}
+2]@sub-masspage,5,U(sub-pageit^${ARG1}^${RECORDED_FILE})gm)))
same  =>   n,ExecIf($[${ISNULL(${ARG2})}=0]?Dial(LOCAL/GO$[${DURATION}
+2]@sub-masspage,5,U(sub-pageit^${ARG2}^${RECORDED_FILE})gm)))
same  =>   n,ExecIf($[${ISNULL(${ARG3})}=0]?Dial(LOCAL/GO$[${DURATION}
+2]@sub-masspage,5,U(sub-pageit^${ARG3}^${RECORDED_FILE})gm)))
same  =>   n,ExecIf($[${ISNULL(${ARG4})}=0]?Dial(LOCAL/GO$[${DURATION}
+2]@sub-masspage,5,U(sub-pageit^${ARG4}^${RECORDED_FILE})gm)))
same  =>   n,ExecIf($[${ISNULL(${ARG5})}=0]?Dial(LOCAL/GO$[${DURATION}
+2]@sub-masspage,5,U(sub-pageit^${ARG5}^${RECORDED_FILE})gm)))
same  =>   n,ExecIf($[${ISNULL(${ARG6})}=0]?Dial(LOCAL/GO$[${DURATION}
+2]@sub-masspage,5,U(sub-pageit^${ARG6}^${RECORDED_FILE})gm)))
same  =>   n,ExecIf($[${ISNULL(${ARG7})}=0]?Dial(LOCAL/GO$[${DURATION}
+2]@sub-masspage,5,U(sub-pageit^${ARG7}^${RECORDED_FILE})gm)))
same  =>   n,ExecIf($[${ISNULL(${ARG8})}=0]?Dial(LOCAL/GO$[${DURATION}
+2]@sub-masspage,5,U(sub-pageit^${ARG8}^${RECORDED_FILE})gm)))
same  =>   n,ExecIf($[${ISNULL(${ARG9})}=0]?Dial(LOCAL/GO$[${DURATION}
+2]@sub-masspage,5,U(sub-pageit^${ARG9}^${RECORDED_FILE})gm)))
same  =>   n,ExecIf($[${ISNULL(${ARG10})}=0]?Dial(LOCAL/GO$[${DURATION}
+2]@sub-masspage,5,U(sub-pageit^${ARG10}^${RECORDED_FILE})gm)))
same  =>   n,Playback(auth-thankyou&goodbye)
same  =>   n,TrySystem(rm -f /var/lib/asterisk/sounds/${RECORDED_FILE}.gsm)
same  =>   n,Hangup

exten => _GO[0-9][0-9],1,NoOP
same  => n,Answer
same  => n,Wait(${EXTEN:-2})
same  => n,Hangup

exten => _GO[0-9],1,NoOP
same  => n,Answer
same  => n,Wait(${EXTEN:-1})
same  => n,Hangup

[sub-pageit]
exten =>  s,1,NoOP
same  =>   n,Set(CALLERID(NAME)=PAGE)
same  =>   n,Set(CALLERID(NUM)=PAGE)
same  =>   n,Page(${ARG1},A(${ARG2})is),10)
same  =>   n,Hangup

;end sub-masspage



On Thu, Dec 5, 2013 at 5:36 PM, John Kiniston wrote:

> I've been working on writing a subroutine to page groups of phones at once
> and I'm having some difficulty.
>
> My goal is to have a user call an extension, I record the page they wish
> to play, I then page out that recorded file to the phones in groups.
>
>
>
> [sub-masspage]
> exten => s,1,NoOP
> same  =>   n,Answer
> same  =>   n,Set(filename=$PAGE)
> same  =>   n,Wait(1)
> same  =>   n,Record(pagequeue/${filename}%d.gsm,0,30,yk)
> same  =>
> n,Set(DURATION=$[CEIL(${STAT(s,/var/lib/asterisk/sounds/${RECORDED_FILE}.gsm)}
> / 1650)])
> same  =>   n,Playback(one-moment-please)
> same  =>   n,Set(MUTEAUDIO(all)=on)
> same  =>   n,SIPAddHeader(Alert-Info: RingAnswer)
> same  =>   n,SipAddHeader(Call-Info: \;answer-after=0)
> ;same  =>   n,Set(TIMEOUT(absolute)=${DURATION})
> same  =>
> n,ExecIf($[${ISNULL(${ARG1})}=0]?Page(${ARG1},n(${RECORDED_FILE})is),10)
> same  =>
> n,ExecIf($[${ISNULL(${ARG2})}=0]?Page(${ARG2},n(${RECORDED_FILE})is),10)
> same  =>
> n,ExecIf($[${ISNULL(${ARG3})}=0]?Page(${ARG3},n(${RECORDED_FILE})is),10)
> same  =>
> n,ExecIf($[${ISNULL(${ARG4})}=0]?Page(${ARG4},n(${RECORDED_FILE})is),10)
> same  =>
> n,ExecIf($[${ISNULL(${ARG5})}=0]?Page(${ARG5},n(${RECORDED_FILE})is),10)
> same  =>
> n,ExecIf($[${ISNULL(${ARG6})}=0]?Page(${ARG6},n(${RECORDED_FILE})is),10)
> same  =>
> n,ExecIf($[${ISNULL(${ARG7})}=0]?Page(${ARG7},n(${RECORDED_FILE})is),10)
> same  =>
> n,ExecIf($[${ISNULL(${ARG8})}=0]?Page(${ARG8},n(${RECORDED_FILE})is),10)
> same  =>
> n,ExecIf($[${ISNULL(${ARG9})}=0]?Page(${ARG9},n(${RECORDED_FILE})is),10)
> same  =>
> n,ExecIf($[${ISNULL(${ARG10})}=0]?Page(${ARG10},n(${RECORDED_FILE})is)10)
> same  =>   n,Set(MUTEAUDIO(all)=off)
> same  =>   n,Playback(goodbye)
> same  =>   n,TrySystem(rm -f /var/lib/asterisk/sounds/${RECORDED_FILE}.gsm)
> same  =>   n,Hangup
> ;end sub-masspage
>
> The issue I'm having is the Page command is putting the phone that's doing
> the paging into the meetme bridge it's creating and the dialplan stops at
> the first Page command.
>
>
> While I was testing it with a single phone I was using Dial instead of Page
>
> same  =>
> n,ExecIf($[${ISNULL(${ARG1})}=0]?Dial(${ARG1},15,A(${RECORDED_FILE})S(${DURATION})mg)
> ))
> same  =>
> n,ExecIf($[${ISNULL(${ARG2})}=0]?Dial(${ARG2},15,A(${RECORDED_FILE})S(${DURATION})mg)))
>
> Which worked great when I sent it one phone in each argument.. Didn't
> think about the fact that the first phone that answers is the only phone
> that

[asterisk-users] Paging in waves.

[John Kiniston]

I've been working on writing a subroutine to page groups of phones at once
and I'm having some difficulty.

My goal is to have a user call an extension, I record the page they wish to
play, I then page out that recorded file to the phones in groups.



[sub-masspage]
exten => s,1,NoOP
same  =>   n,Answer
same  =>   n,Set(filename=$PAGE)
same  =>   n,Wait(1)
same  =>   n,Record(pagequeue/${filename}%d.gsm,0,30,yk)
same  =>
n,Set(DURATION=$[CEIL(${STAT(s,/var/lib/asterisk/sounds/${RECORDED_FILE}.gsm)}
/ 1650)])
same  =>   n,Playback(one-moment-please)
same  =>   n,Set(MUTEAUDIO(all)=on)
same  =>   n,SIPAddHeader(Alert-Info: RingAnswer)
same  =>   n,SipAddHeader(Call-Info: \;answer-after=0)
;same  =>   n,Set(TIMEOUT(absolute)=${DURATION})
same  =>
n,ExecIf($[${ISNULL(${ARG1})}=0]?Page(${ARG1},n(${RECORDED_FILE})is),10)
same  =>
n,ExecIf($[${ISNULL(${ARG2})}=0]?Page(${ARG2},n(${RECORDED_FILE})is),10)
same  =>
n,ExecIf($[${ISNULL(${ARG3})}=0]?Page(${ARG3},n(${RECORDED_FILE})is),10)
same  =>
n,ExecIf($[${ISNULL(${ARG4})}=0]?Page(${ARG4},n(${RECORDED_FILE})is),10)
same  =>
n,ExecIf($[${ISNULL(${ARG5})}=0]?Page(${ARG5},n(${RECORDED_FILE})is),10)
same  =>
n,ExecIf($[${ISNULL(${ARG6})}=0]?Page(${ARG6},n(${RECORDED_FILE})is),10)
same  =>
n,ExecIf($[${ISNULL(${ARG7})}=0]?Page(${ARG7},n(${RECORDED_FILE})is),10)
same  =>
n,ExecIf($[${ISNULL(${ARG8})}=0]?Page(${ARG8},n(${RECORDED_FILE})is),10)
same  =>
n,ExecIf($[${ISNULL(${ARG9})}=0]?Page(${ARG9},n(${RECORDED_FILE})is),10)
same  =>
n,ExecIf($[${ISNULL(${ARG10})}=0]?Page(${ARG10},n(${RECORDED_FILE})is)10)
same  =>   n,Set(MUTEAUDIO(all)=off)
same  =>   n,Playback(goodbye)
same  =>   n,TrySystem(rm -f /var/lib/asterisk/sounds/${RECORDED_FILE}.gsm)
same  =>   n,Hangup
;end sub-masspage

The issue I'm having is the Page command is putting the phone that's doing
the paging into the meetme bridge it's creating and the dialplan stops at
the first Page command.


While I was testing it with a single phone I was using Dial instead of Page

same  =>
n,ExecIf($[${ISNULL(${ARG1})}=0]?Dial(${ARG1},15,A(${RECORDED_FILE})S(${DURATION})mg)
))
same  =>
n,ExecIf($[${ISNULL(${ARG2})}=0]?Dial(${ARG2},15,A(${RECORDED_FILE})S(${DURATION})mg)))

Which worked great when I sent it one phone in each argument.. Didn't think
about the fact that the first phone that answers is the only phone that
answers till I got deeper into writing it... (It's been a long day)

So I'm at a loss as what to do here, Dial almost does what I need other
than the fact that the first phone that answers is the only phone that
bridges, and Page almost does what I need other than my dialplan execution
stops after the first Page command and it's trying to include the phone I'm
calling from as one of the phones it's paging.

I'm using 1.8 so I've got Meetme and Confbridge available but I'm not sure
what to do unless I start playing with LOCAL channels. Maybe I could have
PAGE call Local Channels that have the TIMEOUT set but then I'd need a way
to pass which phones to dial in somehow.

Suggestions?


-- 
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
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Re: [asterisk-users] Paging for Praying

[Steve Edwards]

On Mon, 7 Jan 2013, bilal ghayyad wrote:

Thanks for the help and it seems I deleted some of my emails by mistake! 
I am sorry if I repeated my question.


The lists are archived at:

http://lists.digium.com/pipermail/asterisk-users/

On Wed, 2 Jan 2013, bilal ghayyad wrote:

As I see that the call file is used to generate calls, can I use this 
technique to page the Phones?


Yes. The call file would look something like:

application:page
data:   sip/bilal&sip/steve
channel:local/fajr@prayer-reminder

(This is in alphabetical order. The 'leg' to the channel is executed 
first.)


and a snippet of extensions.conf would look something like:

[prayer-reminder]
exten = fajr,1, verbose(1,[${EXTEN}@${CONTEXT}])
exten = fajr,n, playback(time-for-fajr)
exten = fajr,n, hangup()

When I say Paging, I mean that they are going to hear the sound from the 
speaker (without pickup the handset).


Then you will have to learn how to set the 'auto-answer' SIP header for 
each of your phone types.


The page at:

http://www.voip-info.org/wiki/view/Asterisk+cmd+Page

should help with the specifics of setting auto-answer.

This also means the 'data' line in the call file will change to something 
like:


data:   local/bilal@page&local/steve@page

By using AMI, then I can build PHP script that will use the AMI to do 
the Page?


I'm sorry. I think we had a 'failure to communicate' in your use of the 
word 'page' in your previous emails. I thought you meant 'playback.'


Since you are only placing a single call, the call file approach should be 
fine. But, to answer your question, you should be able to use AMI as well.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Paging unit suggestions

[Pietro Bertera]

Hi,

2013/1/7 Doug Lytle :

> I'm looking for suggestions on a IP based amp or similar that could drive
> the current speakers?  I was envisioning a unit that would register as a SIP
> extension then would handle auto-answer that I could send a sound file to.

I suggest you to take a look on snom PA1:
http://www.snom.com/en/products/sip-paging/snom-pa1/
http://wiki.snom.com/Snom_PA1

this device supports paging via multicast streaming and SIP Alert-Info header.
Is also equipped with 4 I/O pins manageable via DTMF or HTTP commands.

Best regards,
-- 
Bertera Pietro
http://www.bertera.it

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Re: [asterisk-users] Paging for Praying

[Danny Nicholas]

Whether you use .call file or AMI, you should still do the call/page using a
context and that context run the PHP script.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Monday, January 07, 2013 4:30 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Paging for Praying

Thanks for the help and it seems I deleted some of my emails by mistake ! I
am sorry if I repeated my question.

As I see that the call file is used to generate calls, can I use this
technique to page the Phones?

It is one wave file only that need to be Paged for all the Phones connected
on the Asterisk PBX.

When I say Paging, I mean that they are going to hear the sound from the
speaker (without pickup the handset).

By using AMI, then I can build PHP script that will use the AMI to do the
Page?

Thanks and Regards
Bilal

> --
> 

> >> A call file is a text file that you create. The
> format is very
> >> specific.
> 
> On Tue, 1 Jan 2013, bilal ghayyad wrote:
> 
> > * How can I know this format? Because I need to know
> what should I place
> > in this file so it will execute Paging for this group
> of Phones?
> 
> This may help:
> 
>      http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
> 
> >> How many customers will be receiving these
> reminders?
> 
> > * It is required that all the employers at the company
> to hear this on
> > their IP Phones.
> 
> In my experience, you can't just dump xxx call files into the outgoing 
> directory. If you expect more than a dozen or so, you'll have to move 
> them in blocks as they are processed. Another good reason to use AMI.
> 
> >> You can 'schedule' a call file to be processed in
> the future by setting
> >> the file's 'mtime.'
> 
> > * Can you explain for me please?
> 
> Create a file named fajr containing:
> 
>      application:    playback
>      channel:    sip/bilal
>      data:
>     fajr-in-10-minutes
> 
> Copy the file to a directory we assume is on the same file system as
> /var/spool/asterisk/outgoing/:
> 
>      cp\
>          fajr\
>         
> /var/spool/asterisk/tmp/
> 
> Set the file's 'mtime'
> 
>      touch\
>          --date='now + 2
> minutes'\
>          --time=mtime\
>             
> /var/spool/asterisk/tmp/fajr
> 
> Move it to the outgoing directory:
> 
>      mv\
>         
> /var/spool/asterisk/tmp/fajr\
>         
> /var/spool/asterisk/outgoing/
> 
> Your phone should ring in about 2 minutes.
> 
> You may want to look into setting 'auto-answer' or some sort of 
> 'overhead paging' with a very discreet sound file like a short, single 
> beep.
> 
> Please consider AMI if you are looking for a robust service.
> 
> --
> Thanks in advance,
> --
> --- Steve Edwards       sedwa...@sedwards.com
>     Voice: +1-760-468-3867 PST
> Newline


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Re: [asterisk-users] Paging for Praying

[bilal ghayyad]

Thanks for the help and it seems I deleted some of my emails by mistake ! I am 
sorry if I repeated my question.

As I see that the call file is used to generate calls, can I use this technique 
to page the Phones?

It is one wave file only that need to be Paged for all the Phones connected on 
the Asterisk PBX.

When I say Paging, I mean that they are going to hear the sound from the 
speaker (without pickup the handset).

By using AMI, then I can build PHP script that will use the AMI to do the Page?

Thanks and Regards
Bilal

> --
> 

> >> A call file is a text file that you create. The
> format is very 
> >> specific.
> 
> On Tue, 1 Jan 2013, bilal ghayyad wrote:
> 
> > * How can I know this format? Because I need to know
> what should I place 
> > in this file so it will execute Paging for this group
> of Phones?
> 
> This may help:
> 
>      http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
> 
> >> How many customers will be receiving these
> reminders?
> 
> > * It is required that all the employers at the company
> to hear this on 
> > their IP Phones.
> 
> In my experience, you can't just dump xxx call files into
> the outgoing 
> directory. If you expect more than a dozen or so, you'll
> have to move them 
> in blocks as they are processed. Another good reason to use
> AMI.
> 
> >> You can 'schedule' a call file to be processed in
> the future by setting 
> >> the file's 'mtime.'
> 
> > * Can you explain for me please?
> 
> Create a file named fajr containing:
> 
>      application:    playback
>      channel:    sip/bilal
>      data:   
>     fajr-in-10-minutes
> 
> Copy the file to a directory we assume is on the same file
> system as 
> /var/spool/asterisk/outgoing/:
> 
>      cp\
>          fajr\
>         
> /var/spool/asterisk/tmp/
> 
> Set the file's 'mtime'
> 
>      touch\
>          --date='now + 2
> minutes'\
>          --time=mtime\
>             
> /var/spool/asterisk/tmp/fajr
> 
> Move it to the outgoing directory:
> 
>      mv\
>         
> /var/spool/asterisk/tmp/fajr\
>         
> /var/spool/asterisk/outgoing/
> 
> Your phone should ring in about 2 minutes.
> 
> You may want to look into setting 'auto-answer' or some sort
> of 'overhead 
> paging' with a very discreet sound file like a short, single
> beep.
> 
> Please consider AMI if you are looking for a robust
> service.
> 
> -- 
> Thanks in advance,
> -
> Steve Edwards       sedwa...@sedwards.com 
>     Voice: +1-760-468-3867 PST
> Newline             


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Re: [asterisk-users] Paging unit suggestions

[jon pounder]

On 01/07/2013 03:41 PM, Doug Lytle wrote:

The blowing fuses could be related to spikes etc., from a poor 
connection to the source, or a problem with the source hardware.



>> If the amps are good, you could just drive them from a cheap phone 
with a regular headset jack


They aren't, seem to be blowing fuses more often.

Thanks!

Doug


--
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little 
Temporary Safety, deserve neither Liberty nor Safety."



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Re: [asterisk-users] Paging unit suggestions

[Doug Lytle]

>> If the amps are good, you could just drive them from a cheap phone with a 
>> regular headset jack 

They aren't, seem to be blowing fuses more often. 

Thanks! 

Doug 


-- 
Ben Franklin quote: 

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety." 
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Re: [asterisk-users] Paging unit suggestions

[Doug Lytle]

>> you might take a look at Valcom's products. http://www.valcom.com 

I've bookmarked the page, 

Thanks! 

Doug 


-- 
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"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety." 
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Re: [asterisk-users] Paging unit suggestions

[Doug Lytle]

>> the speakers could probably be adapted to work off of any SIP phone 
>> headset/handset 

Interesting thought, but the amp needs to be replaced as well. The 10 year 
Asterisk system will most likely be replaced with a Dell 1U and a Dual-Port 
PRI. 

Thanks for the suggestion! 

Doug 



-- 
Ben Franklin quote: 

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety." 
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Re: [asterisk-users] Paging unit suggestions

[Carlos Alvarez]

On Mon, Jan 7, 2013 at 1:10 PM, Doug Lytle  wrote:

> We currently have an Asterisk system that is hooked up to our old paging
> speakers via sound card, plugged into two amps.
>
> Each amp drives up to 8 analog speakers in each warehouse (we have 2).
> Both warehouses are around 30k square feet.  Both have a large number of
> printing presses.
>
> The computer system is that is running Asterisk is around 10 years old and
> starting to fail.  I'm looking to replace both the system and hopefully
> move to a IP paging system, but wanted to reuse the current speakers.
>

If the amps are good, you could just drive them from a cheap phone with a
regular headset jack.  Set it to auto-answer, send the paging extension to
it.


-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Paging unit suggestions

[Kevin Larsen]

If you want something a little more enterprise ready and tested than a 
RaspberryPi, you might take a look at Valcom's products. 
http://www.valcom.com

We use them for our paging and have been fairly happy with them. Only had 
one small issue that a firmware upgrade took care of.

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208



From:   "Danny Nicholas" 
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 
, 
Date:   01/07/2013 02:14 PM
Subject:Re: [asterisk-users] Paging unit suggestions
Sent by:asterisk-users-boun...@lists.digium.com



From: asterisk-users-boun...@lists.digium.com [
mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Monday, January 07, 2013 2:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Paging unit suggestions
 
We currently have an Asterisk system that is hooked up to our old paging 
speakers via sound card, plugged into two amps.
 
Each amp drives up to 8 analog speakers in each warehouse (we have 2). 
Both warehouses are around 30k square feet.  Both have a large number of 
printing presses.
 
The computer system is that is running Asterisk is around 10 years old and 
starting to fail.  I'm looking to replace both the system and hopefully 
move to a IP paging system, but wanted to reuse the current speakers.
 
I'm looking for suggestions on a IP based amp or similar that could drive 
the current speakers?  I was envisioning a unit that would register as a 
SIP extension then would handle auto-answer that I could send a sound file 
to.
 
Thanks for any help!
 
Doug
 
This may be insane, but it seems from what I read that you could replace 
most 10 year old boxes with a $35 Raspberry Pi.  That not being the case, 
the speakers could probably be adapted to work off of any SIP phone 
headset/handset.--
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Re: [asterisk-users] Paging unit suggestions

[Danny Nicholas]

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Monday, January 07, 2013 2:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Paging unit suggestions

 

We currently have an Asterisk system that is hooked up to our old paging 
speakers via sound card, plugged into two amps.

 

Each amp drives up to 8 analog speakers in each warehouse (we have 2).  Both 
warehouses are around 30k square feet.  Both have a large number of printing 
presses.

 

The computer system is that is running Asterisk is around 10 years old and 
starting to fail.  I'm looking to replace both the system and hopefully move to 
a IP paging system, but wanted to reuse the current speakers.

 

I'm looking for suggestions on a IP based amp or similar that could drive the 
current speakers?  I was envisioning a unit that would register as a SIP 
extension then would handle auto-answer that I could send a sound file to.

 

Thanks for any help!

 

Doug

 

This may be insane, but it seems from what I read that you could replace most 
10 year old boxes with a $35 Raspberry Pi.  That not being the case, the 
speakers could probably be adapted to work off of any SIP phone headset/handset.

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Re: [asterisk-users] Paging unit suggestions

[Christopher Harrington]

On Mon, Jan 7, 2013 at 2:10 PM, Doug Lytle  wrote:

> I'm looking for suggestions on a IP based amp or similar that could drive
> the current speakers?  I was envisioning a unit that would register as a
> SIP extension then would handle auto-answer that I could send a sound file
> to.
>
>
Seems like a RaspberryPi would be a great candidate for this, assuming you
keep the amplifier.


-- 
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248
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[asterisk-users] Paging unit suggestions

[Doug Lytle]

We currently have an Asterisk system that is hooked up to our old paging 
speakers via sound card, plugged into two amps. 

Each amp drives up to 8 analog speakers in each warehouse (we have 2). Both 
warehouses are around 30k square feet. Both have a large number of printing 
presses. 

The computer system is that is running Asterisk is around 10 years old and 
starting to fail. I'm looking to replace both the system and hopefully move to 
a IP paging system, but wanted to reuse the current speakers. 

I'm looking for suggestions on a IP based amp or similar that could drive the 
current speakers? I was envisioning a unit that would register as a SIP 
extension then would handle auto-answer that I could send a sound file to. 

Thanks for any help! 

Doug 

-- 
Ben Franklin quote: 

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety." 
--
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Re: [asterisk-users] Paging for Praying

[Steve Edwards]

Please trim cruft irrelevant to the current questions.

On Wed, 2 Jan 2013, bilal ghayyad wrote:

As I see that the call file is used to generate calls, can I use this 
technique to page the Phones?


Yes. The call file would look something like:

application:page
data:   sip/bilal&sip/steve
channel:local/fajr@prayer-reminder

and a snippet of extensions.conf would look something like:

[prayer-reminder]
exten = fajr,1, verbose(1,[${EXTEN}@${CONTEXT}])
exten = fajr,n, playback(time-for-fajr)
exten = fajr,n, hangup()

When I say Paging, I mean that they are going to hear the sound from the 
speaker (without pickup the handset).


Then you will have to learn how to set the 'auto-answer' SIP header for 
each of your phone types.


The page at:

http://www.voip-info.org/wiki/view/Asterisk+cmd+Page

should help with the specifics of setting auto-answer.

This also means the 'data' line in the call file will change to something 
like:


data:   local/bilal@page&local/steve@page

By using AMI, then I can build PHP script that will use the AMI to do 
the Page?


I'm sorry. I thought we had a 'failure to communicate' in your use of the 
word page in your previous emails. I though you meant playback.


Since you are only placing a single call, the call file approach should be 
fine. But, to answer your question, you should be able to use AMI as well.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Paging for Praying

[bilal ghayyad]

Thanks for the help.

As I see that the call file is used to generate calls, can I use this technique 
to page the Phones?

It is one wave file only that need to be Paged for all the Phones connected on 
the Asterisk PBX.

When I say Paging, I mean that they are going to hear the sound from the 
speaker (without pickup the handset).

By using AMI, then I can build PHP script that will use the AMI to do the Page?

Thanks and Regards
Bilal

> 
> >> A call file is a text file that you create. The
> format is very 
> >> specific.
> 
> On Tue, 1 Jan 2013, bilal ghayyad wrote:
> 
> > * How can I know this format? Because I need to know
> what should I place 
> > in this file so it will execute Paging for this group
> of Phones?
> 
> This may help:
> 
>      http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
> 
> >> How many customers will be receiving these
> reminders?
> 
> > * It is required that all the employers at the company
> to hear this on 
> > their IP Phones.
> 
> In my experience, you can't just dump xxx call files into
> the outgoing 
> directory. If you expect more than a dozen or so, you'll
> have to move them 
> in blocks as they are processed. Another good reason to use
> AMI.
> 
> >> You can 'schedule' a call file to be processed in
> the future by setting 
> >> the file's 'mtime.'
> 
> > * Can you explain for me please?
> 
> Create a file named fajr containing:
> 
>      application:    playback
>      channel:    sip/bilal
>      data:   
>     fajr-in-10-minutes
> 
> Copy the file to a directory we assume is on the same file
> system as 
> /var/spool/asterisk/outgoing/:
> 
>      cp\
>          fajr\
>         
> /var/spool/asterisk/tmp/
> 
> Set the file's 'mtime'
> 
>      touch\
>          --date='now + 2
> minutes'\
>          --time=mtime\
>             
> /var/spool/asterisk/tmp/fajr
> 
> Move it to the outgoing directory:
> 
>      mv\
>         
> /var/spool/asterisk/tmp/fajr\
>         
> /var/spool/asterisk/outgoing/
> 
> Your phone should ring in about 2 minutes.
> 
> You may want to look into setting 'auto-answer' or some sort
> of 'overhead 
> paging' with a very discreet sound file like a short, single
> beep.
> 
> Please consider AMI if you are looking for a robust
> service.

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Re: [asterisk-users] Paging for Praying

[Don Kelly]

Doesn’t the OP wish to page all phones? So it’s not an issue of dumping dozens 
of call files all at once.

 

Does paging work? 

http://www.voip-info.org/wiki/view/Asterisk+cmd+Page

 

http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom

 

Overhead paging might also be something to consider, requiring just one call to 
page “everyone.”

--Don

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Wednesday, January 02, 2013 5:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Paging for Praying

 

How many people do you plan to page? because if numbers are high (or variable) 
you may have an easier life by using some sort of dialer if numbers are not 
very high and two lines are enough, our WombatDialer is free to use.

l.

 

 

2012/12/29 bilal ghayyad 


2) Praying time need to be obtained from text (or database). So, it is not 
always the same time. What actually is needed to be obtained from the text file 
or the database is the time of the pray for each date (for example, if today is 
28 of December so the query will be for this date and then it is required to 
check if the time is same as the current time to page the wave file on the 
Phones).

 

-- 

Loway - home of QueueMetrics - http://queuemetrics.com

Test-drive WombatDialer beta @ http://wombatdialer.com 

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Re: [asterisk-users] Paging for Praying

[Lenz Emilitri]

How many people do you plan to page? because if numbers are high (or
variable) you may have an easier life by using some sort of dialer if
numbers are not very high and two lines are enough, our WombatDialer is
free to use.
l.


2012/12/29 bilal ghayyad 

>
> 2) Praying time need to be obtained from text (or database). So, it is not
> always the same time. What actually is needed to be obtained from the text
> file or the database is the time of the pray for each date (for example, if
> today is 28 of December so the query will be for this date and then it is
> required to check if the time is same as the current time to page the wave
> file on the Phones).
>
>

-- 
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Test-drive WombatDialer beta @ http://wombatdialer.com
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Re: [asterisk-users] Paging for Praying

[Steve Edwards]

On Fri, 28 Dec 2012, Steve Edwards wrote:

A call file is a text file that you create. The format is very 
specific.


On Tue, 1 Jan 2013, bilal ghayyad wrote:

* How can I know this format? Because I need to know what should I place 
in this file so it will execute Paging for this group of Phones?


This may help:

http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out


How many customers will be receiving these reminders?


* It is required that all the employers at the company to hear this on 
their IP Phones.


In my experience, you can't just dump xxx call files into the outgoing 
directory. If you expect more than a dozen or so, you'll have to move them 
in blocks as they are processed. Another good reason to use AMI.


You can 'schedule' a call file to be processed in the future by setting 
the file's 'mtime.'



* Can you explain for me please?


Create a file named fajr containing:

application:playback
channel:sip/bilal
data:   fajr-in-10-minutes

Copy the file to a directory we assume is on the same file system as 
/var/spool/asterisk/outgoing/:


cp\
fajr\
/var/spool/asterisk/tmp/

Set the file's 'mtime'

touch\
--date='now + 2 minutes'\
--time=mtime\
/var/spool/asterisk/tmp/fajr

Move it to the outgoing directory:

mv\
/var/spool/asterisk/tmp/fajr\
/var/spool/asterisk/outgoing/

Your phone should ring in about 2 minutes.

You may want to look into setting 'auto-answer' or some sort of 'overhead 
paging' with a very discreet sound file like a short, single beep.


Please consider AMI if you are looking for a robust service.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Paging for Praying

[bilal ghayyad]

> How many customers will be receiving these reminders?

* It is required that all the employers at the company to hear this on their IP 
Phones.

> What religion is this targeted to?

* Islam.


> A call file is a text file that you create. The format is
> very specific. 

* How can I know this format? Because I need to know what should I place in 
this file so it will execute Paging for this group of Phones?


> You can 
> 'schedule' a call file to be processed in the future by
> setting the file's 
> 'mtime.'

*  Can you explain for me please?

I am fully thanks. 

Regards
Bilal

--

> 
> > I have one more question:
> >
> > What was u meaning by call file and why it is required
> to place them in 
> > the 'astspooldir.'?
> 
> There are 2 methods of originating a call external to
> Asterisk: call files 
> and the Asterisk Manager Interface (AMI).
> 
> A call file is a text file that you create. The format is
> very specific. 
> It could contain (in the context of your needs) the phone
> number to dial 
> and the path of the file to play. A call file is kind of
> like a 'message 
> in a bottle.' You cast it into the sea and hope for the
> best. When this 
> file is "mv'ed" into the directory specified in the Asterisk
> astspooldir 
> variable, Asterisk will read it and try to do what you want.
> You can 
> 'schedule' a call file to be processed in the future by
> setting the file's 
> 'mtime.'
> 
> The Asterisk Manager Interface (AMI) is a TCP connection
> between your 
> program and Asterisk. You can issue commands (like
> originate) and receive 
> responses. AMI is more robust because you can make decisions
> based on the 
> response.
> 
> If robustness is not of primary importance, a script
> scheduled by cron to 
> run after midnight could create the 5 call files for that
> day, setting the 
> 'mtime' of each file before "mv'ing" the file to the
> directory specified 
> by astspooldir -- usually /var/spool/asterisk/outgoing/
> 
> How many customers will be receiving these reminders?
> 
> What religion is this targeted to?

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Re: [asterisk-users] Paging for Praying

[Steve Karmeinsky]

On Sat, Dec 29, 2012 at 12:41:30AM -0500, Steve Totaro wrote:

[snip]
> Wow.  Didn't know there was a rule...  I never got the whole argument.
>  If the flow is top posting, I top post.  If I am certain I can answer
> a simple question with a simple answer, sometimes I will just top
> post.

The first rule of asterisk mailing lists ...

/coat

Steve

-- 
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Re: [asterisk-users] Paging for Praying

[Steve Totaro]

On Fri, Dec 28, 2012 at 11:35 PM, John Novack
 wrote:
>
> Shaun Ruffell wrote:
>>
>> On Fri, Dec 28, 2012 at 06:41:38PM -0800, Steve Edwards wrote:

 On 12/28/2012 08:13 PM, Steve Edwards wrote:

> Please don't top-post. If you don't know what that means, please
> consult Google.
>>>
>>> On Fri, 28 Dec 2012, jon pounder wrote:
>>>
 Please stop saying don't top post, some of us prefer it that way.
>>>
>>> Besides being my preference, it is the documented rule of the
>>> mailing list:
>>>
>>> http://www.asterisk.org/community/discuss/
>>>
>>> Note Mailing List Rules, #5.
>>
>> For a walk down memory lane on top vs bottom posting on the Asterisk
>> mailing lists:
>>
>> http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/254997
>>
> I would add that the "rule # 5 was added long after the first 4, by someone
> in charge after one of the many times this subject has popped up.
>
> Many of the same complainers  routinely do not remove the multi line
> footers, sometimes MANY of them, forcing those who really want to read a
> reply to wade through  the mess. Seems some can't be bothered to delete them
>
> I would also add that rules are made to be broken!
>
> Peg Leg O'Brien
>
>
> --
>
> Dog is my Co-pilot
>

Wow.  Didn't know there was a rule...  I never got the whole argument.
 If the flow is top posting, I top post.  If I am certain I can answer
a simple question with a simple answer, sometimes I will just top
post.

Totally off topic, apologies.  I know I have weighed in once or twice
but really never cared.  If the flow was totally borked and I didn't
care enough to follow the topic, it wasn't that important anyways.

Even more off topic.  Can someone smarter than me get the post totals
for each year?  I was #1 one year.  I am not even talking about
individual post counts though.  It just seems the list has died for
the most part.

Thanks,
Steve T

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Re: [asterisk-users] Paging for Praying

[John Novack]

Shaun Ruffell wrote:

On Fri, Dec 28, 2012 at 06:41:38PM -0800, Steve Edwards wrote:

On 12/28/2012 08:13 PM, Steve Edwards wrote:


Please don't top-post. If you don't know what that means, please
consult Google.

On Fri, 28 Dec 2012, jon pounder wrote:


Please stop saying don't top post, some of us prefer it that way.

Besides being my preference, it is the documented rule of the
mailing list:

http://www.asterisk.org/community/discuss/

Note Mailing List Rules, #5.

For a walk down memory lane on top vs bottom posting on the Asterisk
mailing lists:

http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/254997


I would add that the "rule # 5 was added long after the first 4, by someone in 
charge after one of the many times this subject has popped up.

Many of the same complainers  routinely do not remove the multi line footers, 
sometimes MANY of them, forcing those who really want to read a reply to wade 
through  the mess. Seems some can't be bothered to delete them

I would also add that rules are made to be broken!

Peg Leg O'Brien


--

Dog is my Co-pilot


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Re: [asterisk-users] Paging for Praying

[Shaun Ruffell]

On Fri, Dec 28, 2012 at 06:41:38PM -0800, Steve Edwards wrote:
> >On 12/28/2012 08:13 PM, Steve Edwards wrote:
> >
> >>Please don't top-post. If you don't know what that means, please
> >>consult Google.
> 
> On Fri, 28 Dec 2012, jon pounder wrote:
> 
> >Please stop saying don't top post, some of us prefer it that way.
> 
> Besides being my preference, it is the documented rule of the
> mailing list:
> 
>   http://www.asterisk.org/community/discuss/
> 
> Note Mailing List Rules, #5.

For a walk down memory lane on top vs bottom posting on the Asterisk
mailing lists:

http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/254997

-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Paging for Praying

[Steve Edwards]

On 12/28/2012 08:13 PM, Steve Edwards wrote:

Please don't top-post. If you don't know what that means, please consult 
Google.


On Fri, 28 Dec 2012, jon pounder wrote:


Please stop saying don't top post, some of us prefer it that way.


Besides being my preference, it is the documented rule of the mailing 
list:


http://www.asterisk.org/community/discuss/

Note Mailing List Rules, #5.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Paging for Praying

[jon pounder]

On 12/28/2012 08:13 PM, Steve Edwards wrote:

Please stop saying don't top post, some of us prefer it that way.

Please don't top-post. If you don't know what that means, please 
consult Google.


On Fri, 28 Dec 2012, bilal ghayyad wrote:


I have one more question:

What was u meaning by call file and why it is required to place them 
in the 'astspooldir.'?


There are 2 methods of originating a call external to Asterisk: call 
files and the Asterisk Manager Interface (AMI).


A call file is a text file that you create. The format is very 
specific. It could contain (in the context of your needs) the phone 
number to dial and the path of the file to play. A call file is kind 
of like a 'message in a bottle.' You cast it into the sea and hope for 
the best. When this file is "mv'ed" into the directory specified in 
the Asterisk astspooldir variable, Asterisk will read it and try to do 
what you want. You can 'schedule' a call file to be processed in the 
future by setting the file's 'mtime.'


The Asterisk Manager Interface (AMI) is a TCP connection between your 
program and Asterisk. You can issue commands (like originate) and 
receive responses. AMI is more robust because you can make decisions 
based on the response.


If robustness is not of primary importance, a script scheduled by cron 
to run after midnight could create the 5 call files for that day, 
setting the 'mtime' of each file before "mv'ing" the file to the 
directory specified by astspooldir -- usually 
/var/spool/asterisk/outgoing/


How many customers will be receiving these reminders?

What religion is this targeted to?




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Re: [asterisk-users] Paging for Praying

[Steve Edwards]

Please don't top-post. If you don't know what that means, please consult 
Google.


On Fri, 28 Dec 2012, bilal ghayyad wrote:


I have one more question:

What was u meaning by call file and why it is required to place them in 
the 'astspooldir.'?


There are 2 methods of originating a call external to Asterisk: call files 
and the Asterisk Manager Interface (AMI).


A call file is a text file that you create. The format is very specific. 
It could contain (in the context of your needs) the phone number to dial 
and the path of the file to play. A call file is kind of like a 'message 
in a bottle.' You cast it into the sea and hope for the best. When this 
file is "mv'ed" into the directory specified in the Asterisk astspooldir 
variable, Asterisk will read it and try to do what you want. You can 
'schedule' a call file to be processed in the future by setting the file's 
'mtime.'


The Asterisk Manager Interface (AMI) is a TCP connection between your 
program and Asterisk. You can issue commands (like originate) and receive 
responses. AMI is more robust because you can make decisions based on the 
response.


If robustness is not of primary importance, a script scheduled by cron to 
run after midnight could create the 5 call files for that day, setting the 
'mtime' of each file before "mv'ing" the file to the directory specified 
by astspooldir -- usually /var/spool/asterisk/outgoing/


How many customers will be receiving these reminders?

What religion is this targeted to?

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Paging for Praying

[bilal ghayyad]

Dear Steve;

Thanks a lot for your help.

Specifically what I need is the following:

1) One wave file to be paged at the Phones. In the 5 times, the same file will 
be used.

2) Praying time need to be obtained from text (or database). So, it is not 
always the same time. What actually is needed to be obtained from the text file 
or the database is the time of the pray for each date (for example, if today is 
28 of December so the query will be for this date and then it is required to 
check if the time is same as the current time to page the wave file on the 
Phones).

I have one more question:

What was u meaning by call file and why it is required to place them in the 
'astspooldir.'?

Regards
Bilal

> Please don't top-post.
> 
> On Thu, Dec 27, 2012 at 1:29 PM, bilal ghayyad 
> 
> wrote:
> 
> How can I have Paging on Asterisk to call for pray?
> 
> The pray is 5 times in a day and there is a timing for pray
> (actually it 
> can be existed in a text file or database for the next 2 or
> 5 years).
> 
> [mailto:asterisk-users-boun...@lists.digium.com]
> On Behalf Of Bharat 
> Lalcheta
> 
> However, its easy to build a script in php or perl or any
> other language 
> which check time from file or database and generate call
> file which 
> execute paging in asterisk. Just put this script in cron.
> Thats it...
> 
> From: Danny Nicholas 
> 
> I would set up 5 shell files called pray1.sh, pray2.sh, etc
> and then set 
> up 5 entries in /etc/crontab to run them at the specified
> time daily. The 
> file pray1.sh should look something like this:
> 
> #!/bin/sh
> cp /pray1/*.call /tmp
> mv /tmp/*.call /var/spool/asterisk/outgoing
> 
> the entry in /etc/crontab would look like this
> 
> 0 8 *** root /usr/bin/pray1.sh
> 
> This would run pray1.sh at 8 am daily.
> 
> On Thu, 27 Dec 2012, bilal ghayyad wrote:
> 
> > Thanks a lot for your kindly reply and help.
> >
> > Really I did not understand why you need to place them
> in the 
> > /var/spool/asterisk/outgoing?
> 
> The appropriate solution needs a lot more detail to be
> useful.
> 
> Is this just to remind you or is this the foundation of a
> new product for 
> thousands of customers?
> 
> Is there a message or verse associated with each of the 5
> reminders or is 
> 'time to pray' sufficient?
> 
> Is there a penalty associated with missing a prayer like
> eternal 
> damnation? (AMI is more robust than call files.)
> 
> The answers would help guide you in deciding if a simple
> cron based shell 
> script generating call files or a database driven AMI daemon
> is the best 
> approach.
> 
> In answer to your specific question, the call files need to
> be "mv'ed"
> into /v/s/a/o/ because:
> 
> ) You need to use mv instead of cp because mv is an 'atomic'
> function* 
> meaning it happens all at once so that Asterisk will not try
> to read an 
> incomplete file.
> 
> ) This is the default value of 'astspooldir.' You can
> specify a different 
> location in asterisk.conf if needed.
> 
> *) Assuming the source and destination are on the same
> filesystem.

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Re: [asterisk-users] Paging for Praying

[Steve Edwards]

Please don't top-post.

On Thu, Dec 27, 2012 at 1:29 PM, bilal ghayyad  
wrote:


How can I have Paging on Asterisk to call for pray?

The pray is 5 times in a day and there is a timing for pray (actually it 
can be existed in a text file or database for the next 2 or 5 years).


[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bharat 
Lalcheta


However, its easy to build a script in php or perl or any other language 
which check time from file or database and generate call file which 
execute paging in asterisk. Just put this script in cron. Thats it...


From: Danny Nicholas 

I would set up 5 shell files called pray1.sh, pray2.sh, etc and then set 
up 5 entries in /etc/crontab to run them at the specified time daily. The 
file pray1.sh should look something like this:


#!/bin/sh
cp /pray1/*.call /tmp
mv /tmp/*.call /var/spool/asterisk/outgoing

the entry in /etc/crontab would look like this

0 8 *** root /usr/bin/pray1.sh

This would run pray1.sh at 8 am daily.

On Thu, 27 Dec 2012, bilal ghayyad wrote:


Thanks a lot for your kindly reply and help.

Really I did not understand why you need to place them in the 
/var/spool/asterisk/outgoing?


The appropriate solution needs a lot more detail to be useful.

Is this just to remind you or is this the foundation of a new product for 
thousands of customers?


Is there a message or verse associated with each of the 5 reminders or is 
'time to pray' sufficient?


Is there a penalty associated with missing a prayer like eternal 
damnation? (AMI is more robust than call files.)


The answers would help guide you in deciding if a simple cron based shell 
script generating call files or a database driven AMI daemon is the best 
approach.


In answer to your specific question, the call files need to be "mv'ed"
into /v/s/a/o/ because:

) You need to use mv instead of cp because mv is an 'atomic' function* 
meaning it happens all at once so that Asterisk will not try to read an 
incomplete file.


) This is the default value of 'astspooldir.' You can specify a different 
location in asterisk.conf if needed.


*) Assuming the source and destination are on the same filesystem.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Paging for Praying

[bilal ghayyad]

Thanks a lot for your kindly reply and help.

Really I did not understand why you need to place them in the 
/var/spool/asterisk/outgoing?

Regards
Bilal


---
> 
> I would set up 5 shell files called pray1.sh, pray2.sh, etc
> and then set up
> 5 entries in /etc/crontab to run them at the specified time
> daily.  The file
> pray1.sh should look something like this:
> 
> #!/bin/sh
> 
> cp /pray1/*.call /tmp
> 
> mv /tmp/*.call /var/spool/asterisk/outgoing
> 
>  
> 
> the entry in /etc/crontab would look like this
> 
> 0 8 *** root /usr/bin/pray1.sh
> 
>  
> 
> This would run pray1.sh at 8 am daily.
> 
>  
> 
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com]
> On Behalf Of Bharat
> Lalcheta
> Sent: Thursday, December 27, 2012 2:22 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Paging for Praying
> 
>  
> 
> I dont think this is existed.
> 
>  
> 
> However, its easy to build a script in php or perl or any
> other language
> which check time from file or database and generate call
> file which execute
> paging in asterisk. Just put this script in cron. Thats
> it...
> 
>  
> 
> Regards,
> 
>  
> 
> Bharat Lalcheta
> 
>  
> 
> 
> 
>  
> 
> On Thu, Dec 27, 2012 at 1:29 PM, bilal ghayyad 
> wrote:
> 
> Hello;
> 
> How can I have Paging on Asterisk to call for pray?
> 
> The pray is 5 times in a day and there is a timing for pray
> (actually it can
> be existed in a text file or database for the next 2 or 5
> years).
> 
> My question is compound from two parts:
> 
> How can I have Automatic Page?
> 
> The automatic page should happens by reading the time and
> check if the time
> is same as this time, then do the Page. How? Is it by cron?
> 
> Someone told me that do a cron that call a script which will
> check the time,
> if the time came to do th Page, then do a Page. But really I
> do not know how
> this can be done and I do not know if this is already
> existed?
> 
> Regards
> Bilal


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Re: [asterisk-users] Paging for Praying

[Danny Nicholas]

I would set up 5 shell files called pray1.sh, pray2.sh, etc and then set up
5 entries in /etc/crontab to run them at the specified time daily.  The file
pray1.sh should look something like this:

#!/bin/sh

cp /pray1/*.call /tmp

mv /tmp/*.call /var/spool/asterisk/outgoing

 

the entry in /etc/crontab would look like this

0 8 *** root /usr/bin/pray1.sh

 

This would run pray1.sh at 8 am daily.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bharat
Lalcheta
Sent: Thursday, December 27, 2012 2:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Paging for Praying

 

I dont think this is existed.

 

However, its easy to build a script in php or perl or any other language
which check time from file or database and generate call file which execute
paging in asterisk. Just put this script in cron. Thats it...

 

Regards,

 

Bharat Lalcheta

 



 

On Thu, Dec 27, 2012 at 1:29 PM, bilal ghayyad  wrote:

Hello;

How can I have Paging on Asterisk to call for pray?

The pray is 5 times in a day and there is a timing for pray (actually it can
be existed in a text file or database for the next 2 or 5 years).

My question is compound from two parts:

How can I have Automatic Page?

The automatic page should happens by reading the time and check if the time
is same as this time, then do the Page. How? Is it by cron?

Someone told me that do a cron that call a script which will check the time,
if the time came to do th Page, then do a Page. But really I do not know how
this can be done and I do not know if this is already existed?

Regards
Bilal

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Bharat Lalcheta 

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Re: [asterisk-users] Paging for Praying

[Bharat Lalcheta]

I dont think this is existed.

However, its easy to build a script in php or perl or any other language
which check time from file or database and generate call file which execute
paging in asterisk. Just put this script in cron. Thats it...

Regards,

Bharat Lalcheta




On Thu, Dec 27, 2012 at 1:29 PM, bilal ghayyad  wrote:

> Hello;
>
> How can I have Paging on Asterisk to call for pray?
>
> The pray is 5 times in a day and there is a timing for pray (actually it
> can be existed in a text file or database for the next 2 or 5 years).
>
> My question is compound from two parts:
>
> How can I have Automatic Page?
>
> The automatic page should happens by reading the time and check if the
> time is same as this time, then do the Page. How? Is it by cron?
>
> Someone told me that do a cron that call a script which will check the
> time, if the time came to do th Page, then do a Page. But really I do not
> know how this can be done and I do not know if this is already existed?
>
> Regards
> Bilal
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



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[asterisk-users] Paging for Praying

[bilal ghayyad]

Hello;

How can I have Paging on Asterisk to call for pray?

The pray is 5 times in a day and there is a timing for pray (actually it can be 
existed in a text file or database for the next 2 or 5 years).

My question is compound from two parts:

How can I have Automatic Page?

The automatic page should happens by reading the time and check if the time is 
same as this time, then do the Page. How? Is it by cron?

Someone told me that do a cron that call a script which will check the time, if 
the time came to do th Page, then do a Page. But really I do not know how this 
can be done and I do not know if this is already existed?

Regards
Bilal

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Re: [asterisk-users] Paging with Polycom 3.3.x

[Dave Fullerton]

Actually, I don't think that has been the case for quite a while. Anyone 
can get the latest firmware directly from polycom. Including, 3.3.1F


http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html

On 02/24/2011 03:32 PM, Mike wrote:

Sorry, I realize my tone might not go down well.   I didn't mean to sound
like a jerk, but I was just stating that resellers are also authorized to
distribute the firmware to their customers if I recall correctly, so
everybody can get the firmware for free, just not directly from Polycom.



And I don't actually think this is the best way for Polycom to do things,
but that`s the way things are.



Mike





From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, February 24, 2011 3:27 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Paging with Polycom 3.3.x



Polycom are at 3.3.1 now, so 3.3.0 should be fair game.



It has nothing to do with paying or not, the company that sold you the phone
should be able to give you the latest version no?  Unless you bought from a
guy who found a box that fell off a truck.or some third-rate reseller.



Mike



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell
Sent: Thursday, February 24, 2011 3:21 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Paging with Polycom 3.3.x



Is 3.3.x downloadable for non-paying people yet?







From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, February 24, 2011 2:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Paging with Polycom 3.3.x



Thank you Terry and Ryan, I will try those things and see if I can find my
problem. Will definitely come back with my solution in case it helps
somebody else.



Mike





Hi Terry,



I did that, and did find a autoRingAnswer and Ringanswer modes that I tried
to use, but somewhere I am missing something that breaks it. If ever you
find what you did, I`d appreciate if you'd share with me.



Mike





  

and



where the timeout is the ampount of time on milliseconds before it goes to
speaker.



These values are in the sip.cfg, so in your server it may be sip_316.cfg.





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Re: [asterisk-users] Paging with Polycom 3.3.x

[Mike]

Sorry, I realize my tone might not go down well.   I didn't mean to sound
like a jerk, but I was just stating that resellers are also authorized to
distribute the firmware to their customers if I recall correctly, so
everybody can get the firmware for free, just not directly from Polycom.

 

And I don't actually think this is the best way for Polycom to do things,
but that`s the way things are.

 

Mike

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, February 24, 2011 3:27 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Paging with Polycom 3.3.x

 

Polycom are at 3.3.1 now, so 3.3.0 should be fair game.

 

It has nothing to do with paying or not, the company that sold you the phone
should be able to give you the latest version no?  Unless you bought from a
guy who found a box that fell off a truck.or some third-rate reseller.

 

Mike

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell
Sent: Thursday, February 24, 2011 3:21 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Paging with Polycom 3.3.x

 

Is 3.3.x downloadable for non-paying people yet?

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, February 24, 2011 2:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Paging with Polycom 3.3.x

 

Thank you Terry and Ryan, I will try those things and see if I can find my
problem. Will definitely come back with my solution in case it helps
somebody else.

 

Mike

 

 

Hi Terry,

 

I did that, and did find a autoRingAnswer and Ringanswer modes that I tried
to use, but somewhere I am missing something that breaks it. If ever you
find what you did, I`d appreciate if you'd share with me.

 

Mike

 

 

 

and



where the timeout is the ampount of time on milliseconds before it goes to
speaker.

 

These values are in the sip.cfg, so in your server it may be sip_316.cfg.

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Re: [asterisk-users] Paging with Polycom 3.3.x

[Mike]

Polycom are at 3.3.1 now, so 3.3.0 should be fair game.

 

It has nothing to do with paying or not, the company that sold you the phone
should be able to give you the latest version no?  Unless you bought from a
guy who found a box that fell off a truck.or some third-rate reseller.

 

Mike

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell
Sent: Thursday, February 24, 2011 3:21 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Paging with Polycom 3.3.x

 

Is 3.3.x downloadable for non-paying people yet?

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, February 24, 2011 2:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Paging with Polycom 3.3.x

 

Thank you Terry and Ryan, I will try those things and see if I can find my
problem. Will definitely come back with my solution in case it helps
somebody else.

 

Mike

 

 

Hi Terry,

 

I did that, and did find a autoRingAnswer and Ringanswer modes that I tried
to use, but somewhere I am missing something that breaks it. If ever you
find what you did, I`d appreciate if you'd share with me.

 

Mike

 

 

 

and



where the timeout is the ampount of time on milliseconds before it goes to
speaker.

 

These values are in the sip.cfg, so in your server it may be sip_316.cfg.

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Re: [asterisk-users] Paging with Polycom 3.3.x

[William Stillwell]

Is 3.3.x downloadable for non-paying people yet?

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, February 24, 2011 2:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Paging with Polycom 3.3.x

 

Thank you Terry and Ryan, I will try those things and see if I can find my
problem. Will definitely come back with my solution in case it helps
somebody else.

 

Mike

 

 

Hi Terry,

 

I did that, and did find a autoRingAnswer and Ringanswer modes that I tried
to use, but somewhere I am missing something that breaks it. If ever you
find what you did, I`d appreciate if you'd share with me.

 

Mike

 

 

 

and



where the timeout is the ampount of time on milliseconds before it goes to
speaker.

 

These values are in the sip.cfg, so in your server it may be sip_316.cfg.

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Re: [asterisk-users] Paging with Polycom 3.3.x

[Mike]

Thank you Terry and Ryan, I will try those things and see if I can find my
problem. Will definitely come back with my solution in case it helps
somebody else.

 

Mike

 

 

Hi Terry,

 

I did that, and did find a autoRingAnswer and Ringanswer modes that I tried
to use, but somewhere I am missing something that breaks it. If ever you
find what you did, I`d appreciate if you'd share with me.

 

Mike

 

 

 

and



where the timeout is the ampount of time on milliseconds before it goes to
speaker.

 

These values are in the sip.cfg, so in your server it may be sip_316.cfg.

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Re: [asterisk-users] Paging with Polycom 3.3.x

[Terry Brummell]

From: asterisk-users-boun...@lists.digium.com on behalf of Mike
Sent: Thu 2/24/2011 2:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Paging with Polycom 3.3.x



Hi Terry,

 

I did that, and did find a autoRingAnswer and Ringanswer modes that I tried to 
use, but somewhere I am missing something that breaks it. If ever you find what 
you did, I`d appreciate if you'd share with me.

 

Mike

 

 

 

and



where the timeout is the ampount of time on milliseconds before it goes to 
speaker.

 

These values are in the sip.cfg, so in your server it may be sip_316.cfg.

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Re: [asterisk-users] Paging with Polycom 3.3.x

[Ryan Wagoner]

On Thu, Feb 24, 2011 at 1:41 PM, Mike  wrote:
> Hi,
>
>
>
> My phones stopped auto-answering when being paged, since I moved on to
> Polycom firmware 3.3.0 (3.3.1 is the same, I tried).  That is with Asterisk
> 1.6.2.16.
>
>
>
> I looked at the wiki but nothing I try there works, even if I cut and paste
> the same setup.
>
>
>
> Any one has any idea of what I should change from my old 3.2.3 setup?  My
> older phone (501) still using 3.1.6 still auto-answer correctly.
>

Polycom changed some of the config file options as outlined in the UC
Software upgrade guide. I am using the following for paging.


  
  
 
  

Ryan

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Re: [asterisk-users] Paging with Polycom 3.3.x

[Terry Brummell]

 



From: asterisk-users-boun...@lists.digium.com on behalf of Mike
Sent: Thu 2/24/2011 2:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Paging with Polycom 3.3.x



Hi Terry,

 

I did that, and did find a autoRingAnswer and Ringanswer modes that I tried to 
use, but somewhere I am missing something that breaks it. If ever you find what 
you did, I`d appreciate if you'd share with me.

 

Mike

 

Looking for it now

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Re: [asterisk-users] Paging with Polycom 3.3.x

[Mike]

Hi Terry,

 

I did that, and did find a autoRingAnswer and Ringanswer modes that I tried
to use, but somewhere I am missing something that breaks it. If ever you
find what you did, I`d appreciate if you'd share with me.

 

Mike

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell
Sent: Thursday, February 24, 2011 2:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Paging with Polycom 3.3.x

 

If you compare a working config with a non-working you will see something
with the answer type.  I had that issue until I down rev'd.  Look for
something like "Ring Answer", I forget the exact details now.

 

  _  

From: asterisk-users-boun...@lists.digium.com on behalf of Mike
Sent: Thu 2/24/2011 1:41 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Paging with Polycom 3.3.x

Hi,

 

My phones stopped auto-answering when being paged, since I moved on to
Polycom firmware 3.3.0 (3.3.1 is the same, I tried).  That is with Asterisk
1.6.2.16.

 

I looked at the wiki but nothing I try there works, even if I cut and paste
the same setup.

 

Any one has any idea of what I should change from my old 3.2.3 setup?  My
older phone (501) still using 3.1.6 still auto-answer correctly.

 

Regards,

 

Mike

 

 

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Re: [asterisk-users] Paging with Polycom 3.3.x

[Terry Brummell]

If you compare a working config with a non-working you will see something with 
the answer type.  I had that issue until I down rev'd.  Look for something like 
"Ring Answer", I forget the exact details now.



From: asterisk-users-boun...@lists.digium.com on behalf of Mike
Sent: Thu 2/24/2011 1:41 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Paging with Polycom 3.3.x



Hi,

 

My phones stopped auto-answering when being paged, since I moved on to Polycom 
firmware 3.3.0 (3.3.1 is the same, I tried).  That is with Asterisk 1.6.2.16.

 

I looked at the wiki but nothing I try there works, even if I cut and paste the 
same setup.

 

Any one has any idea of what I should change from my old 3.2.3 setup?  My older 
phone (501) still using 3.1.6 still auto-answer correctly.

 

Regards,

 

Mike

 

 

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[asterisk-users] Paging with Polycom 3.3.x

[Mike]

Hi,

 

My phones stopped auto-answering when being paged, since I moved on to
Polycom firmware 3.3.0 (3.3.1 is the same, I tried).  That is with Asterisk
1.6.2.16.

 

I looked at the wiki but nothing I try there works, even if I cut and paste
the same setup.

 

Any one has any idea of what I should change from my old 3.2.3 setup?  My
older phone (501) still using 3.1.6 still auto-answer correctly.

 

Regards,

 

Mike

 

 

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Re: [asterisk-users] Paging a message. How?

[Doug Lytle]

Russell Brown wrote:

I can do a normal page (phones auto-answer on speaker) with SipAddHeader
but that doesn't let me play a pre-recorded message.

Any suggestions?
   

Dial plan:

[polycom-page]

;*
;* Play the previously recorded page file over the
;* Polycom speaker.
;*


exten => s,1,NoCDR()
exten => s,n,ResetCDR()
exten => s,n,Set(CALLERID(name)=Paging)
exten => s,n,Set(CALLERID(num)=44)
exten => s,n,Wait(.5)
exten => s,n,Playback(out)
exten => s,n,Hangup()


Call file:

cat 0.call

Channel: Local/4220@sip/n
CallerID: Paging <44>
WaitTime: 5
MaxRetries: 0
Context: polycom-page
Extension: s
Priority: 1


Doug


--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


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Re: [asterisk-users] Paging a message. How?

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russell Brown
Sent: Tuesday, February 15, 2011 12:40 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Paging a message. How?


I'm scratching my head trying to work out a way of sending a
pre-recorded message as a 'Page' to a list of phones ( "Oi!  you muppets
you've left the server room door open!" or somesuch message :-)
controlled by an external trigger.

I can do a normal page (phones auto-answer on speaker) with SipAddHeader
but that doesn't let me play a pre-recorded message.

Any suggestions?


-- 
 Regards,
 Russell

Set up a context to play the message like this
[playit]
Exten => s,1,playback(${ARG1})
Exten => s,n,playback(vm-goodbye)
Exten => s,n,hangup

Then add this to your "normal page" function 
Exten => s,n,Goto(playit,s,1)



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[asterisk-users] Paging a message. How?

[Russell Brown]

I'm scratching my head trying to work out a way of sending a
pre-recorded message as a 'Page' to a list of phones ( "Oi!  you muppets
you've left the server room door open!" or somesuch message :-)
controlled by an external trigger.

I can do a normal page (phones auto-answer on speaker) with SipAddHeader
but that doesn't let me play a pre-recorded message.

Any suggestions?


-- 
 Regards,
 Russell
 
| Russell Brown  | MAIL: russ...@lls.com PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com  |
| Peterborough, England  | WWW Play: http://www.ruffle.me.uk |
 

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[asterisk-users] paging/intercom

[lists]

 

 

I'm having hard times with paging intercom

 

Heres my dialplan

 

exten => 777,1,Goto(intercom,777,1)

 

[intercom]

exten => 777,1,SIPAddHeader(Call-Info: \;answer-after=0)

exten => 777,2,Page(Local/3...@page& Local/3...@page& Local/3...@page)

 

 

[page] ; Paging context

exten => _X.,1,Macro(page,SIP/${EXTEN})

 

[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,SIPAddHeader(Alert-Info: Ring Answer) ; this one is for the
Polycom IP601

exten => s,3,Dial(${ARG1}|3|) ; should ring 3 seconds

exten => s,4,Hangup

exten => s,104,Hangup

 

 

 

 

 

 

Problem is that if lets say 310 is on the phone with a client.. and one
pages all.. (777) then the 310 phone (Linksys 942)  puts current call on
hold and or drops the call to answer page.

 

Is that the send audio to speaker option in preference of the phone that's
not right ?

 

 

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[asterisk-users] Paging with Pickup

[Asaf Ben Aroch]

Hi,

I'm trying to achieve the following feature that's common in Avaya
systems:

 

A user page all extensions in a full duplex mode- he can hear all, and
all can hear him via their phones' speaker. When one of the extensions
picks up the handset, the call is bridged between the pager and the
person who picked up. All the rest are disconnected.

 

Does anyone have an idea or ever implemented such behavior? I was
thinking about adding an extra step for the picking up user to dial **
and do a dynamic feature that will run a script to kick all the rest
from the Meetme of the Page().

 

Thoughts?

 

A

 

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Re: [asterisk-users] Paging, Polycom and whispers

[Danny Nicholas]

You can send an IM to the phone with a text message.  Assuming that the
phone has more than 1 line and at least one is open, the call should go
through without effecting the existing call.  To do this from the dialplan,
you could set up something like this:

Exten => 411,1,Dial(SIP/100,1)
Exten => Sendtext(You have a call on park 701)
Exten -> hangup(}

This also assumes that the polycom has presence enabled.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton
Sent: Tuesday, December 02, 2008 10:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Paging, Polycom and whispers

Mike wrote:
> Hi,
> 
>  
> 
> Is there a way to page a Polycom phone that is already in use (if, of
> course, the call isn't on speakerphone already)?
> 

I've never been able to find a way. Any attempt I made either put the 
existing call on hold to auto-answer the page or the page just rang at 
the phone and then caused other issues.

I'm not sure you'll have any luck with other SIP phones either. What 
you're asking it to do is accept two simultaneous calls but put each 
call on a different listening device (handset/speakerphone in this case).

The closest you might get is to rig a dialplan that would use chanspy in 
whisper mode to play the page through the current audio device if the 
phone is busy. I don't know how to go about doing that however.

-Dave

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Re: [asterisk-users] Paging, Polycom and whispers

[Dave Fullerton]

Mike wrote:
> Hi,
> 
>  
> 
> Is there a way to page a Polycom phone that is already in use (if, of
> course, the call isn't on speakerphone already)?
> 

I've never been able to find a way. Any attempt I made either put the 
existing call on hold to auto-answer the page or the page just rang at 
the phone and then caused other issues.

I'm not sure you'll have any luck with other SIP phones either. What 
you're asking it to do is accept two simultaneous calls but put each 
call on a different listening device (handset/speakerphone in this case).

The closest you might get is to rig a dialplan that would use chanspy in 
whisper mode to play the page through the current audio device if the 
phone is busy. I don't know how to go about doing that however.

-Dave

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[asterisk-users] Paging, Polycom and whispers

[Mike]

Hi,

 

Is there a way to page a Polycom phone that is already in use (if, of
course, the call isn't on speakerphone already)?

 

 

 

Mike

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Re: [asterisk-users] Paging

[Alex Balashov]

bilal ghayyad wrote:
> Hi List;
> 
> In the below link:
> 
> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
> 
> I saw this line and did not find for it explaination,
> anyone can explain it?
> 
> exten =>
> 7999,2,Page(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED]&Local/[EMAIL 
> PROTECTED]/n&Local/interal
> [EMAIL PROTECTED]|)
> 
> What local means? and why to use @page?
> What Local/[EMAIL PROTECTED]/n means?
> What Local/interal [EMAIL PROTECTED]| means?

The question you are asking is less about paging and more about "local 
channels," which are a piece of dial plan architecture:

http://www.voip-info.org/wiki/view/Asterisk+local+channels

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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[asterisk-users] Paging

[bilal ghayyad]

Hi List;

In the below link:

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page

I saw this line and did not find for it explaination,
anyone can explain it?

exten =>
7999,2,Page(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED]&Local/[EMAIL 
PROTECTED]/n&Local/interal
[EMAIL PROTECTED]|)

What local means? and why to use @page?
What Local/[EMAIL PROTECTED]/n means?
What Local/interal [EMAIL PROTECTED]| means?

Any help?
Regards
Bilal


  

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Re: [asterisk-users] Paging for analoge devices

[bilal ghayyad]

Hi Steve;

If we give a look for the link
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
there are some topics not cleared to complete the
paging senario:

1) Why to use Set(_ALERT_INFO="RA")? And how can I
know what each device take? 

2) Does the following work for Polycom:

SIPAddHeader(Call-Info:\;answer-after=0)


3) Can I do paging without use Page and only by using
Dial (but I will set the previous paramters as
needed)?

4) Any configuration need to be done on the Meet Me
conference to have the Paging working fine?

Regards
Bilal

--- Steve Totaro <[EMAIL PROTECTED]>
wrote:

> Bilal,
> 
> Providing your Asterisk box has onboard sound or you
> can add a card or
> even USB sound then you will just use your Asterisk
> server to act as a
> phone basically.  It even has autoanswer so it
> should be perfect.
> 
> I think you have enough options now to act.
> 
> http://www.voip-info.org/wiki-Asterisk+tips+console
> 
> Then you need to either feed it into an AMP (Bogen
> or whatever) or buy
> speakers with built in AMPs and volume control.
> 
> Thanks,
> Steve Totaro
> 
> 
> On Tue, Apr 29, 2008 at 3:43 AM, bilal ghayyad
> <[EMAIL PROTECTED]> wrote:
> > Dear Steve;
> >
> >  Using the computer sound card is a very nice
> solution,
> >  but how to connect it to the pbx? How to connect
> it to
> >  the fxs port and give it an extension? Do I need
> an
> >  phone?
> >
> >
> >
> >  Regards
> >  Bilal
> >  --- Steve Totaro <[EMAIL PROTECTED]>
> >  wrote:
> >
> >  > Bilal,
> >  >
> >  > Geez, yes, that is how overhead paging works. 
> You
> >  > can even buy the
> >  > speakers with built in AMPs and volume control
> and
> >  > you could use your
> >  > computer's sound card.
> >  >
> >  > Thanks,
> >  > Steve Totaro
> >  >
> >  > On Mon, Apr 28, 2008 at 6:30 PM, bilal ghayyad
> >  > <[EMAIL PROTECTED]> wrote:
> >  > > I will try to see the analoge and how it can
> be
> >  > done.
> >  > >
> >  > >  About the overhead, do u mean the AMP and
> >  > speakers or
> >  > >  what?
> >  > >
> >  > >
> >  > >
> >  > >  Regards
> >  > >  Bilal
> >  > >  --- Steve Totaro
> <[EMAIL PROTECTED]>
> >  > >  wrote:
> >  > >
> >  > >  > Bilal,
> >  > >  >
> >  > >  > So you want to page through your analog
> phones,
> >  > no
> >  > >  > overhead paging.  I
> >  > >  > doubt this is possible.
> >  > >  >
> >  > >  > I think your options are an IP phones such
> as
> >  > the
> >  > >  > Polycom that
> >  > >  > supports paging or using an AMP and
> speakers
> >  > usually
> >  > >  > mounted high on
> >  > >  > the wall or flush with a tile ceiling.
> >  > >  >
> >  > >  > Thanks,
> >  > >  > Steve Totaro
> >  > >  >
> >  > >  > On Mon, Apr 28, 2008 at 11:58 AM, bilal
> ghayyad
> >  > >  > <[EMAIL PROTECTED]> wrote:
> >  > >  > > Dear Steve;
> >  > >  > >
> >  > >  > >  Already we need to use the same analoge
> >  > phones
> >  > >  > that is
> >  > >  > >  connected to the fxs ports, to do
> paging to
> >  > it.
> >  > >  > >
> >  > >  > >  What overhead?
> >  > >  > >
> >  > >  > >
> >  > >  > >
> >  > >  > >  Regards
> >  > >  > >  Bilal
> >  > >  > >  --- Steve Totaro
> >  > <[EMAIL PROTECTED]>
> >  > >  > >  wrote:
> >  > >  > >
> >  > >  > >  > Bilal,
> >  > >  > >  >
> >  > >  > >  > You will have great luck with Polycom
> >  > then, not
> >  > >  > sure
> >  > >  > >  > about others.
> >  > >  > >  > Why not go for overhead paging?  It
> will
> >  > be
> >  > >  > much
> >  > >  > >  > easier.
> >  > >  > >  >
> >  > >  > >  > Thanks,
> >  > >  > >  > Steve Totaro
> >  > >  > >  >
> >  > >  > >  > On Mon, Apr 28, 2008 at 9:24 AM,
> bilal
> >  > ghayyad
> >  > >  > >  > <[EMAIL PROTECTED]> wrote:
> >  > >  > >  > > And auto answer also will be needed
> for
> >  > IP
> >  > >  > Phones?
> >  > >  > >  > >
> >  > >  > >  > >
> >  > >  > >  > >
> >  > >  > >  > >  Regards
> >  > >  > >  > >  Bilal
> >  > >  > >  > >  --- Steve Totaro
> >  > >  > <[EMAIL PROTECTED]>
> >  > >  > >  > >  wrote:
> >  > >  > >  > >
> >  > >  > >  > >  > Bilal,
> >  > >  > >  > >  >
> >  > >  > >  > >  > Sorry to reply to my reply but
> if you
> >  > can
> >  > >  > >  > register
> >  > >  > >  > >  > multiple accounts
> >  > >  > >  > >  > and setup auto answer on one of
> those
> >  > >  > accounts,
> >  > >  > >  > it
> >  > >  > >  > >  > could work.  The
> >  > >  > >  > >  > problem is, I am not sure if
> there
> >  > are any
> >  > >  > ATAs
> >  > >  > >  > that
> >  > >  > >  > >  > have this
> >  > >  > >  > >  > ability.
> >  > >  > >  > >  >
> >  > >  > >  > >  > Thanks,
> >  > >  > >  > >  > Steve Totaro
> >  > >  > >  > >  >
> >  > >  > >  > >  > On Mon, Apr 28, 2008 at 9:06 AM,
> >  > Steve
> >  > >  > Totaro
> >  > >  > >  > >  > <[EMAIL PROTECTED]>
> >  > wrote:
> >  > >  > >  > >  > > Bilal,
> >  > >  > >  > >  > >
> >  > >  > >  > >  > >  No, I do not think that you
> can
> >  > make
> >  > >  > this
> >  > >  > >  > work.
> >  > >  > >  > >  > You would obviously
> >  > >  > >  > >  > >  need auto answer or how els

Re: [asterisk-users] Paging for analoge devices

[Alexander Lopez]

I am going on memory but I do recall that Aastra had a phone that used
ADSI codes that would 'turn on' a speaker on an analog phone

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Steve Totaro
> Sent: Monday, April 28, 2008 3:10 PM
> To: bilal ghayyad; Asterisk Users Mailing List - Non-Commercial
Discussion
> Subject: Re: [asterisk-users] Paging for analoge devices
> 
> Bilal,
> 
> So you want to page through your analog phones, no overhead paging.  I
> doubt this is possible.
> 
> I think your options are an IP phones such as the Polycom that
> supports paging or using an AMP and speakers usually mounted high on
> the wall or flush with a tile ceiling.
> 
> Thanks,
> Steve Totaro
> 
> On Mon, Apr 28, 2008 at 11:58 AM, bilal ghayyad <[EMAIL PROTECTED]>
> wrote:
> > Dear Steve;
> >
> >  Already we need to use the same analoge phones that is
> >  connected to the fxs ports, to do paging to it.
> >
> >  What overhead?
> >
> >
> >
> >  Regards
> >  Bilal
> >  --- Steve Totaro <[EMAIL PROTECTED]>
> >  wrote:
> >
> >  > Bilal,
> >  >
> >  > You will have great luck with Polycom then, not sure
> >  > about others.
> >  > Why not go for overhead paging?  It will be much
> >  > easier.
> >  >
> >  > Thanks,
> >  > Steve Totaro
> >  >
> >  > On Mon, Apr 28, 2008 at 9:24 AM, bilal ghayyad
> >  > <[EMAIL PROTECTED]> wrote:
> >  > > And auto answer also will be needed for IP Phones?
> >  > >
> >  > >
> >  > >
> >  > >  Regards
> >  > >  Bilal
> >  > >  --- Steve Totaro <[EMAIL PROTECTED]>
> >  > >  wrote:
> >  > >
> >  > >  > Bilal,
> >  > >  >
> >  > >  > Sorry to reply to my reply but if you can
> >  > register
> >  > >  > multiple accounts
> >  > >  > and setup auto answer on one of those accounts,
> >  > it
> >  > >  > could work.  The
> >  > >  > problem is, I am not sure if there are any ATAs
> >  > that
> >  > >  > have this
> >  > >  > ability.
> >  > >  >
> >  > >  > Thanks,
> >  > >  > Steve Totaro
> >  > >  >
> >  > >  > On Mon, Apr 28, 2008 at 9:06 AM, Steve Totaro
> >  > >  > <[EMAIL PROTECTED]> wrote:
> >  > >  > > Bilal,
> >  > >  > >
> >  > >  > >  No, I do not think that you can make this
> >  > work.
> >  > >  > You would obviously
> >  > >  > >  need auto answer or how else would the audio
> >  > come
> >  > >  > out of the speakers?
> >  > >  > >
> >  > >  > >  I thought you were talking about overhead
> >  > paging.
> >  > >  > >
> >  > >  > >  Thanks,
> >  > >  > >  Steve Totaro
> >  > >  > >
> >  > >  > >
> >  > >  > >
> >  > >  > >  On Mon, Apr 28, 2008 at 8:49 AM, bilal
> >  > ghayyad
> >  > >  > <[EMAIL PROTECTED]> wrote:
> >  > >  > >  > Dear Steve;
> >  > >  > >  >
> >  > >  > >  >  I was asking about the ability to do
> >  > paging on
> >  > >  > the
> >  > >  > >  >  normal analoge phones (which is connected
> >  > to
> >  > >  > fxs
> >  > >  > >  >  ports, and it already receive and
> >  > originate
> >  > >  > calls on
> >  > >  > >  >  the system). So does that work? If yes,
> >  > do I
> >  > >  > need
> >  > >  > >  >  auto-answer? If I need auto-answer here,
> >  > then
> >  > >  > I will
> >  > >  > >  >  face a problem in the calls which coming
> >  > >  > normally to
> >  > >  > >  >  the analoge phone (when someone dial that
> >  > >  > extension).
> >  > >  > >  >
> >  > >  > >  >  Any advise?
> >  > >  > >  >
> >  > >  > >  >  Regards
> >  > >  > >  >  Bilal
> >  > >  > >  >  --- Steve Totaro
> >  > >  > <[EMAIL PROTECTED]>
> >  > >  > >  >  wrote:
> >

Re: [asterisk-users] Paging for analoge devices

[Steve Totaro]

Bilal,

So you want to page through your analog phones, no overhead paging.  I
doubt this is possible.

I think your options are an IP phones such as the Polycom that
supports paging or using an AMP and speakers usually mounted high on
the wall or flush with a tile ceiling.

Thanks,
Steve Totaro

On Mon, Apr 28, 2008 at 11:58 AM, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> Dear Steve;
>
>  Already we need to use the same analoge phones that is
>  connected to the fxs ports, to do paging to it.
>
>  What overhead?
>
>
>
>  Regards
>  Bilal
>  --- Steve Totaro <[EMAIL PROTECTED]>
>  wrote:
>
>  > Bilal,
>  >
>  > You will have great luck with Polycom then, not sure
>  > about others.
>  > Why not go for overhead paging?  It will be much
>  > easier.
>  >
>  > Thanks,
>  > Steve Totaro
>  >
>  > On Mon, Apr 28, 2008 at 9:24 AM, bilal ghayyad
>  > <[EMAIL PROTECTED]> wrote:
>  > > And auto answer also will be needed for IP Phones?
>  > >
>  > >
>  > >
>  > >  Regards
>  > >  Bilal
>  > >  --- Steve Totaro <[EMAIL PROTECTED]>
>  > >  wrote:
>  > >
>  > >  > Bilal,
>  > >  >
>  > >  > Sorry to reply to my reply but if you can
>  > register
>  > >  > multiple accounts
>  > >  > and setup auto answer on one of those accounts,
>  > it
>  > >  > could work.  The
>  > >  > problem is, I am not sure if there are any ATAs
>  > that
>  > >  > have this
>  > >  > ability.
>  > >  >
>  > >  > Thanks,
>  > >  > Steve Totaro
>  > >  >
>  > >  > On Mon, Apr 28, 2008 at 9:06 AM, Steve Totaro
>  > >  > <[EMAIL PROTECTED]> wrote:
>  > >  > > Bilal,
>  > >  > >
>  > >  > >  No, I do not think that you can make this
>  > work.
>  > >  > You would obviously
>  > >  > >  need auto answer or how else would the audio
>  > come
>  > >  > out of the speakers?
>  > >  > >
>  > >  > >  I thought you were talking about overhead
>  > paging.
>  > >  > >
>  > >  > >  Thanks,
>  > >  > >  Steve Totaro
>  > >  > >
>  > >  > >
>  > >  > >
>  > >  > >  On Mon, Apr 28, 2008 at 8:49 AM, bilal
>  > ghayyad
>  > >  > <[EMAIL PROTECTED]> wrote:
>  > >  > >  > Dear Steve;
>  > >  > >  >
>  > >  > >  >  I was asking about the ability to do
>  > paging on
>  > >  > the
>  > >  > >  >  normal analoge phones (which is connected
>  > to
>  > >  > fxs
>  > >  > >  >  ports, and it already receive and
>  > originate
>  > >  > calls on
>  > >  > >  >  the system). So does that work? If yes,
>  > do I
>  > >  > need
>  > >  > >  >  auto-answer? If I need auto-answer here,
>  > then
>  > >  > I will
>  > >  > >  >  face a problem in the calls which coming
>  > >  > normally to
>  > >  > >  >  the analoge phone (when someone dial that
>  > >  > extension).
>  > >  > >  >
>  > >  > >  >  Any advise?
>  > >  > >  >
>  > >  > >  >  Regards
>  > >  > >  >  Bilal
>  > >  > >  >  --- Steve Totaro
>  > >  > <[EMAIL PROTECTED]>
>  > >  > >  >  wrote:
>  > >  > >  >
>  > >  > >  >
>  > >  > >  >
>  > >  > >  >  > Bogen is the defacto standard in
>  > quality
>  > >  > paging
>  > >  > >  >  > equipment.
>  > >  > >  >  >
>  > >  > >  >  > FXS port should work just fine with
>  > >  > auto-answer
>  > >  > >  >  > using any AMP.
>  > >  > >  >  >
>  > >  > >  >  > The Onkyo suggestion was just an idea
>  > for a
>  > >  > cheaper
>  > >  > >  >  > but probably
>  > >  > >  >  > higher quality audio with more volume
>  > in a
>  > >  > loud
>  > >  > >  >  > environment.  I was
>  > >  > >  >  > not saying you "needed" to use Onkyo,
>  > just
>  > >  > that you
>  > >  > >  >  > could build a
>  > >  > >  >  > paging system with a regular tuner,
>  > possibly
>  > >  > have
>  > >  > >  >  > better sound, and do
>  > >  > >  >  > it cheaper than going Bogen.
>  > >  > >  >  >
>  > >  > >  >  > Thanks,
>  > >  > >  >  > Steve Totaro
>  > >  > >  >  >
>  > >  > >  >  > On Mon, Apr 28, 2008 at 6:39 AM, bilal
>  > >  > ghayyad
>  > >  > >  >  > <[EMAIL PROTECTED]> wrote:
>  > >  > >  >  > > Hi All;
>  > >  > >  >  > >
>  > >  > >  >  > >  Just would like to know if Boger,
>  > UPAM,
>  > >  > and Onkyo
>  > >  > >  >  > can
>  > >  > >  >  > >  be connected to the FXS ports and
>  > take
>  > >  > extension
>  > >  > >  >  > from
>  > >  > >  >  > >  asterisk?
>  > >  > >  >  > >
>  > >  > >  >  > >  And why Onkyo is needed if I need to
>  > use
>  > >  > horn
>  > >  > >  >  > >  speakers?
>  > >  > >  >  > >
>  > >  > >  >  > >  Regards
>  > >  > >  >  > >  Bilal
>  > >  > >  >  > >
>  > >  > >  >  > >  ---
>  > >  > >  >  > >
>  > >  > >  >  > >  >  >  > bilal ghayyad wrote:
>  > >  > >  >  > >  >  >
>  > >  > >  >  > >  >  > > > Steve Totaro wrote:
>  > >  > >  >  > >  >  >  > > > On 4/5/08, bilal ghayyad
>  > >  > >  >  > >   <[EMAIL PROTECTED]> wrote:
>  > >  > >  >  > >  >  >
>  > >  > >  >  > >  >  > > > > > Anyone knows (tried) to
>  > use
>  > >  > Page for
>  > >  > >  >  > >   analoge
>  > >  > >  >  > >  >  >  > > > > phone(zaptel channel -
>  > >  > fxs)? If
>  > >  > >  >  > yes,
>  > >  > >  >  > >   how?
>  > >  > >  >  > >  >  >  > > >
>  > >  > >

Re: [asterisk-users] Paging for analoge devices

[Steve Totaro]

On Mon, Apr 28, 2008 at 10:28 AM, John Faubion <[EMAIL PROTECTED]> wrote:
> > > need auto answer or how else would the audio come out of
>  > the speakers?
>
>  One of the lowest cost solutions I've used for this is a Grandstream BT-200.
>  You can buy the phone for around $50, it supports auto answer, and it has a
>  miniature audio jack already on it. Just run the output to an amplifier and
>  speakers and your good to go.
>
>  John
>

Yes I suspect the GS ATAs (286/486) have the same option and have a
simple single pair analog RJ11 jack for an analog device.  Might be
even cheaper and definitely lower profile than a full blown phone.

Besides that I have had bad luck with GS phones (based on the BT102s)
losing registration.  This was quite some time ago but the ATAs never
gave me any problems.

Thanks,
Steve Totaro

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Re: [asterisk-users] Paging for analoge devices

[John Faubion]

> > need auto answer or how else would the audio come out of 
> the speakers?

One of the lowest cost solutions I've used for this is a Grandstream BT-200.
You can buy the phone for around $50, it supports auto answer, and it has a
miniature audio jack already on it. Just run the output to an amplifier and
speakers and your good to go.

John


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Re: [asterisk-users] Paging for analoge devices

[Steve Totaro]

Bilal,

Sorry to reply to my reply but if you can register multiple accounts
and setup auto answer on one of those accounts, it could work.  The
problem is, I am not sure if there are any ATAs that have this
ability.

Thanks,
Steve Totaro

On Mon, Apr 28, 2008 at 9:06 AM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
> Bilal,
>
>  No, I do not think that you can make this work.  You would obviously
>  need auto answer or how else would the audio come out of the speakers?
>
>  I thought you were talking about overhead paging.
>
>  Thanks,
>  Steve Totaro
>
>
>
>  On Mon, Apr 28, 2008 at 8:49 AM, bilal ghayyad <[EMAIL PROTECTED]> wrote:
>  > Dear Steve;
>  >
>  >  I was asking about the ability to do paging on the
>  >  normal analoge phones (which is connected to fxs
>  >  ports, and it already receive and originate calls on
>  >  the system). So does that work? If yes, do I need
>  >  auto-answer? If I need auto-answer here, then I will
>  >  face a problem in the calls which coming normally to
>  >  the analoge phone (when someone dial that extension).
>  >
>  >  Any advise?
>  >
>  >  Regards
>  >  Bilal
>  >  --- Steve Totaro <[EMAIL PROTECTED]>
>  >  wrote:
>  >
>  >
>  >
>  >  > Bogen is the defacto standard in quality paging
>  >  > equipment.
>  >  >
>  >  > FXS port should work just fine with auto-answer
>  >  > using any AMP.
>  >  >
>  >  > The Onkyo suggestion was just an idea for a cheaper
>  >  > but probably
>  >  > higher quality audio with more volume in a loud
>  >  > environment.  I was
>  >  > not saying you "needed" to use Onkyo, just that you
>  >  > could build a
>  >  > paging system with a regular tuner, possibly have
>  >  > better sound, and do
>  >  > it cheaper than going Bogen.
>  >  >
>  >  > Thanks,
>  >  > Steve Totaro
>  >  >
>  >  > On Mon, Apr 28, 2008 at 6:39 AM, bilal ghayyad
>  >  > <[EMAIL PROTECTED]> wrote:
>  >  > > Hi All;
>  >  > >
>  >  > >  Just would like to know if Boger, UPAM, and Onkyo
>  >  > can
>  >  > >  be connected to the FXS ports and take extension
>  >  > from
>  >  > >  asterisk?
>  >  > >
>  >  > >  And why Onkyo is needed if I need to use horn
>  >  > >  speakers?
>  >  > >
>  >  > >  Regards
>  >  > >  Bilal
>  >  > >
>  >  > >  ---
>  >  > >
>  >  > >  >  >  > bilal ghayyad wrote:
>  >  > >  >  >
>  >  > >  >  > > > Steve Totaro wrote:
>  >  > >  >  >  > > > On 4/5/08, bilal ghayyad
>  >  > >   <[EMAIL PROTECTED]> wrote:
>  >  > >  >  >
>  >  > >  >  > > > > > Anyone knows (tried) to use Page for
>  >  > >   analoge
>  >  > >  >  >  > > > > phone(zaptel channel - fxs)? If
>  >  > yes,
>  >  > >   how?
>  >  > >  >  >  > > >
>  >  > >  >  >  > > > Bogen Rulez
>  >  > >  >  >
>  >  > >  >  > > >
>  >  > >  >  >  > > Sorry I did not understand any thing
>  >  > from
>  >  > >  your
>  >  > >   reply.
>  >  > >  >  >  >
>  >  > >  >  >
>  >  > >  >  > > Google is your friend.  I discovered very
>  >  > >  quickly what
>  >  > >   they were
>  >  > >   talking
>  >  > >  >  >  > about by googling.
>  >  > >  >  >
>  >  > >  >  >  You may be able to find the device a little
>  >  > >  cheaper, but
>  >  > >   without
>  >  > >   the brand
>  >  > >  >  >  name, by searching for "UPAM".  Typically,
>  >  > these
>  >  > >  will have
>  >  > >   the
>  >  > >   "AT&T" or
>  >  > >  >  >  "Lucent" names branded on them, despite
>  >  > being
>  >  > >  identical and
>  >  > >  >  >  manufactured by exactly the same entity.
>  >  > >  >  >
>  >  > >  >  >  --
>  >  > >  >  >  Tilghman
>  >  > >  >  >
>  >  > >  >
>  >  > >  >  Yes, but Bogen sounds so much better than UPAM
>  >  > ;-)
>  >  > >  >
>  >  > >  >  (noted for future paging needs, I am doing a
>  >  > >  concrete factory
>  >  > >   next
>  >  > >   moth)
>  >  > >  >
>  >  > >  >  Thanks,
>  >  > >  >  Steve Totaro
>  >  > >  >
>  >  > >
>  >  > >  Just thinking about it, I could pick up a nice
>  >  > >  refurbed Onkyo and some
>  >  > >  horn speakers and create an industrial paging
>  >  > system
>  >  > >  complete with a
>  >  > >  couple of zones for much less than any other over
>  >  > >  priced "paging
>  >  > >  solution".
>  >  > >
>  >  > >  A and B speakers could be used for two zones and
>  >  > >  balance could be used
>  >  > >  for two other zones.
>  >  > >
>  >  > >
>  >  >
>  >  
> http://www.ecost.com/detail.aspx?edp=40236464&source=EWBBASE&ci_src=17588969&ci_sku=40236464
>  >  > >
>  >  > >  Thanks,
>  >  > >  Steve
>  >  > >
>  >  > >
>  >  > >
>  >  > >
>  >  > >
>  >  > >
>  >  > >
>  >  >
>  >  
> 
>  >  > >  Be a better friend, newshound, and
>  >  > >  know-it-all with Yahoo! Mobile.  Try it now.
>  >  >
>  >  http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ
>  >  > >
>  >  >
>  >
>  >
>  >
>  >   
> 
>  >  Be a better friend, newshound, and
>  >  know-it-all with Yahoo! Mobile. 

Re: [asterisk-users] Paging for analoge devices

[Steve Totaro]

Bilal,

No, I do not think that you can make this work.  You would obviously
need auto answer or how else would the audio come out of the speakers?

I thought you were talking about overhead paging.

Thanks,
Steve Totaro

On Mon, Apr 28, 2008 at 8:49 AM, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> Dear Steve;
>
>  I was asking about the ability to do paging on the
>  normal analoge phones (which is connected to fxs
>  ports, and it already receive and originate calls on
>  the system). So does that work? If yes, do I need
>  auto-answer? If I need auto-answer here, then I will
>  face a problem in the calls which coming normally to
>  the analoge phone (when someone dial that extension).
>
>  Any advise?
>
>  Regards
>  Bilal
>  --- Steve Totaro <[EMAIL PROTECTED]>
>  wrote:
>
>
>
>  > Bogen is the defacto standard in quality paging
>  > equipment.
>  >
>  > FXS port should work just fine with auto-answer
>  > using any AMP.
>  >
>  > The Onkyo suggestion was just an idea for a cheaper
>  > but probably
>  > higher quality audio with more volume in a loud
>  > environment.  I was
>  > not saying you "needed" to use Onkyo, just that you
>  > could build a
>  > paging system with a regular tuner, possibly have
>  > better sound, and do
>  > it cheaper than going Bogen.
>  >
>  > Thanks,
>  > Steve Totaro
>  >
>  > On Mon, Apr 28, 2008 at 6:39 AM, bilal ghayyad
>  > <[EMAIL PROTECTED]> wrote:
>  > > Hi All;
>  > >
>  > >  Just would like to know if Boger, UPAM, and Onkyo
>  > can
>  > >  be connected to the FXS ports and take extension
>  > from
>  > >  asterisk?
>  > >
>  > >  And why Onkyo is needed if I need to use horn
>  > >  speakers?
>  > >
>  > >  Regards
>  > >  Bilal
>  > >
>  > >  ---
>  > >
>  > >  >  >  > bilal ghayyad wrote:
>  > >  >  >
>  > >  >  > > > Steve Totaro wrote:
>  > >  >  >  > > > On 4/5/08, bilal ghayyad
>  > >   <[EMAIL PROTECTED]> wrote:
>  > >  >  >
>  > >  >  > > > > > Anyone knows (tried) to use Page for
>  > >   analoge
>  > >  >  >  > > > > phone(zaptel channel - fxs)? If
>  > yes,
>  > >   how?
>  > >  >  >  > > >
>  > >  >  >  > > > Bogen Rulez
>  > >  >  >
>  > >  >  > > >
>  > >  >  >  > > Sorry I did not understand any thing
>  > from
>  > >  your
>  > >   reply.
>  > >  >  >  >
>  > >  >  >
>  > >  >  > > Google is your friend.  I discovered very
>  > >  quickly what
>  > >   they were
>  > >   talking
>  > >  >  >  > about by googling.
>  > >  >  >
>  > >  >  >  You may be able to find the device a little
>  > >  cheaper, but
>  > >   without
>  > >   the brand
>  > >  >  >  name, by searching for "UPAM".  Typically,
>  > these
>  > >  will have
>  > >   the
>  > >   "AT&T" or
>  > >  >  >  "Lucent" names branded on them, despite
>  > being
>  > >  identical and
>  > >  >  >  manufactured by exactly the same entity.
>  > >  >  >
>  > >  >  >  --
>  > >  >  >  Tilghman
>  > >  >  >
>  > >  >
>  > >  >  Yes, but Bogen sounds so much better than UPAM
>  > ;-)
>  > >  >
>  > >  >  (noted for future paging needs, I am doing a
>  > >  concrete factory
>  > >   next
>  > >   moth)
>  > >  >
>  > >  >  Thanks,
>  > >  >  Steve Totaro
>  > >  >
>  > >
>  > >  Just thinking about it, I could pick up a nice
>  > >  refurbed Onkyo and some
>  > >  horn speakers and create an industrial paging
>  > system
>  > >  complete with a
>  > >  couple of zones for much less than any other over
>  > >  priced "paging
>  > >  solution".
>  > >
>  > >  A and B speakers could be used for two zones and
>  > >  balance could be used
>  > >  for two other zones.
>  > >
>  > >
>  >
>  
> http://www.ecost.com/detail.aspx?edp=40236464&source=EWBBASE&ci_src=17588969&ci_sku=40236464
>  > >
>  > >  Thanks,
>  > >  Steve
>  > >
>  > >
>  > >
>  > >
>  > >
>  > >
>  > >
>  >
>  
> 
>  > >  Be a better friend, newshound, and
>  > >  know-it-all with Yahoo! Mobile.  Try it now.
>  >
>  http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ
>  > >
>  >
>
>
>
>   
> 
>  Be a better friend, newshound, and
>  know-it-all with Yahoo! Mobile.  Try it now.  
> http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ
>

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[asterisk-users] Paging for analoge devices

[bilal ghayyad]

Hi All;

Just would like to know if Boger, UPAM, and Onkyo can
be connected to the FXS ports and take extension from
asterisk?

And why Onkyo is needed if I need to use horn
speakers? 

Regards
Bilal

---

>  >  > bilal ghayyad wrote:
>  >
>  > > > Steve Totaro wrote:
>  >  > > > On 4/5/08, bilal ghayyad
 <[EMAIL PROTECTED]> wrote:
>  >
>  > > > > > Anyone knows (tried) to use Page for
 analoge
>  >  > > > > phone(zaptel channel - fxs)? If yes,
 how?
>  >  > > >
>  >  > > > Bogen Rulez
>  >
>  > > >
>  >  > > Sorry I did not understand any thing from
your
 reply.
>  >  >
>  >
>  > > Google is your friend.  I discovered very
quickly what
 they were
 talking
>  >  > about by googling.
>  >
>  >  You may be able to find the device a little
cheaper, but
 without
 the brand
>  >  name, by searching for "UPAM".  Typically, these
will have
 the
 "AT&T" or
>  >  "Lucent" names branded on them, despite being
identical and
>  >  manufactured by exactly the same entity.
>  >
>  >  --
>  >  Tilghman
>  >
>
>  Yes, but Bogen sounds so much better than UPAM ;-)
>
>  (noted for future paging needs, I am doing a
concrete factory
 next
 moth)
>
>  Thanks,
>  Steve Totaro
>

Just thinking about it, I could pick up a nice
refurbed Onkyo and some
horn speakers and create an industrial paging system
complete with a
couple of zones for much less than any other over
priced "paging
solution".

A and B speakers could be used for two zones and
balance could be used
for two other zones.

http://www.ecost.com/detail.aspx?edp=40236464&source=EWBBASE&ci_src=17588969&ci_sku=40236464

Thanks,
Steve





  

Be a better friend, newshound, and 
know-it-all with Yahoo! Mobile.  Try it now.  
http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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Re: [asterisk-users] Paging for analoge devices

[Jay R. Ashworth]

On Mon, Apr 07, 2008 at 08:14:41AM -0400, Steve Totaro wrote:
> (noted for future paging needs, I am doing a concrete factory next moth)

I hear *excellent* things about (and from :-) SoundSphere paging
speakers.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

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Re: [asterisk-users] Paging for analoge devices

[Steve Totaro]

On Mon, Apr 7, 2008 at 8:14 AM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
> On Mon, Apr 7, 2008 at 8:09 AM, Tilghman Lesher
>  <[EMAIL PROTECTED]> wrote:
>  > On Monday 07 April 2008 05:15:42 Rob Hillis wrote:
>  >  > bilal ghayyad wrote:
>  >
>  > > > Steve Totaro wrote:
>  >  > > > On 4/5/08, bilal ghayyad <[EMAIL PROTECTED]> wrote:
>  >
>  > > > > > Anyone knows (tried) to use Page for analoge
>  >  > > > > phone(zaptel channel - fxs)? If yes, how?
>  >  > > >
>  >  > > > Bogen Rulez
>  >
>  > > >
>  >  > > Sorry I did not understand any thing from your reply.
>  >  >
>  >
>  > > Google is your friend.  I discovered very quickly what they were talking
>  >  > about by googling.
>  >
>  >  You may be able to find the device a little cheaper, but without the brand
>  >  name, by searching for "UPAM".  Typically, these will have the "AT&T" or
>  >  "Lucent" names branded on them, despite being identical and
>  >  manufactured by exactly the same entity.
>  >
>  >  --
>  >  Tilghman
>  >
>
>  Yes, but Bogen sounds so much better than UPAM ;-)
>
>  (noted for future paging needs, I am doing a concrete factory next moth)
>
>  Thanks,
>  Steve Totaro
>

Just thinking about it, I could pick up a nice refurbed Onkyo and some
horn speakers and create an industrial paging system complete with a
couple of zones for much less than any other over priced "paging
solution".

A and B speakers could be used for two zones and balance could be used
for two other zones.

http://www.ecost.com/detail.aspx?edp=40236464&source=EWBBASE&ci_src=17588969&ci_sku=40236464

Thanks,
Steve

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Re: [asterisk-users] Paging for analoge devices

[Steve Totaro]

On Mon, Apr 7, 2008 at 8:09 AM, Tilghman Lesher
<[EMAIL PROTECTED]> wrote:
> On Monday 07 April 2008 05:15:42 Rob Hillis wrote:
>  > bilal ghayyad wrote:
>
> > > Steve Totaro wrote:
>  > > > On 4/5/08, bilal ghayyad <[EMAIL PROTECTED]> wrote:
>
> > > > > Anyone knows (tried) to use Page for analoge
>  > > > > phone(zaptel channel - fxs)? If yes, how?
>  > > >
>  > > > Bogen Rulez
>
> > >
>  > > Sorry I did not understand any thing from your reply.
>  >
>
> > Google is your friend.  I discovered very quickly what they were talking
>  > about by googling.
>
>  You may be able to find the device a little cheaper, but without the brand
>  name, by searching for "UPAM".  Typically, these will have the "AT&T" or
>  "Lucent" names branded on them, despite being identical and
>  manufactured by exactly the same entity.
>
>  --
>  Tilghman
>

Yes, but Bogen sounds so much better than UPAM ;-)

(noted for future paging needs, I am doing a concrete factory next moth)

Thanks,
Steve Totaro

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Re: [asterisk-users] Paging for analoge devices

[Tilghman Lesher]

On Monday 07 April 2008 05:15:42 Rob Hillis wrote:
> bilal ghayyad wrote:
> > Steve Totaro wrote:
> > > On 4/5/08, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> > > > Anyone knows (tried) to use Page for analoge
> > > > phone(zaptel channel - fxs)? If yes, how?
> > >
> > > Bogen Rulez
> >
> > Sorry I did not understand any thing from your reply.
>
> Google is your friend.  I discovered very quickly what they were talking
> about by googling.

You may be able to find the device a little cheaper, but without the brand
name, by searching for "UPAM".  Typically, these will have the "AT&T" or
"Lucent" names branded on them, despite being identical and
manufactured by exactly the same entity.

-- 
Tilghman

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Re: [asterisk-users] Paging for analoge devices

[Rob Hillis]

Google is your friend.  I discovered very quickly what they were talking 
about by googling.



bilal ghayyad wrote:

Dear Steve & Doug;

Sorry I did not understand any thing from your reply.

---

Bogen Rulez

On 4/5/08, bilal ghayyad <[EMAIL PROTECTED]> wrote:
  

Hi;

Anyone knows (tried) to use Page for analoge
phone(zaptel channel - fxs)? If yes, how?

Regards
Bilal









Steve Totaro wrote:
  

Bogen Rulez
  



That it does!


Regards
Bilal


  

You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost.  
http://tc.deals.yahoo.com/tc/blockbuster/text5.com


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Re: [asterisk-users] Paging for analoge devices

[bilal ghayyad]

Dear Steve & Doug;

Sorry I did not understand any thing from your reply.

---

Bogen Rulez

On 4/5/08, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> Hi;
>
> Anyone knows (tried) to use Page for analoge
> phone(zaptel channel - fxs)? If yes, how?
>
> Regards
> Bilal
>
>
>
>



Steve Totaro wrote:
> Bogen Rulez
>   

That it does!


Regards
Bilal


  

You rock. That's why Blockbuster's offering you one month of Blockbuster Total 
Access, No Cost.  
http://tc.deals.yahoo.com/tc/blockbuster/text5.com

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Re: [asterisk-users] Paging for analoge devices

[Doug Lytle]

Steve Totaro wrote:
> Bogen Rulez
>   

That it does!



-- 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] Paging for analoge devices

[Steve Totaro]

Bogen Rulez

On 4/5/08, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> Hi;
>
> Anyone knows (tried) to use Page for analoge
> phone(zaptel channel - fxs)? If yes, how?
>
> Regards
> Bilal
>
>
>
> 
> You rock. That's why Blockbuster's offering you one month of Blockbuster
> Total Access, No Cost.
> http://tc.deals.yahoo.com/tc/blockbuster/text5.com
>
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[asterisk-users] Paging for analoge devices

[bilal ghayyad]

Hi;

Anyone knows (tried) to use Page for analoge
phone(zaptel channel - fxs)? If yes, how?

Regards
Bilal


  

You rock. That's why Blockbuster's offering you one month of Blockbuster Total 
Access, No Cost.  
http://tc.deals.yahoo.com/tc/blockbuster/text5.com

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[asterisk-users] Paging and conferences/chan_alsa.

[Thomas Kenyon]

Okay, so this time I wont send the email to the bounce address :-)

Can I include an alsa channel in a Page() channel list?

Is there a way I can have an extension call a group of phones and put 
them into a pre-existing conference (muted) while the caller goes into 
the conference (unmuted, and preferably with a beep announcement)?

TIA for any clarifications.

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[asterisk-users] Paging Recording File

[Forrest Beck]

I am looking to see if anyone has seen this problem before.  I am  
setting the MEETME_RECORDINGFILE variable in a macro, then using the r  
option with the Page application to record the page.  But the page is  
only recorded to the file specified in  MEETME_RECORDINGFILE  
sometimes...  Sometimes it works and sometimes it doesn't.  When it  
doesn't work it places the recorded file in the sounds dir with a  
meetme-conf-. name.  Here is my Macro.

Basically it is getting my phones that begin with a certain number  
from the realtime database to create a variable with a value that ='s  
SIP/6001&SIP/6002&SIP/6003  this is passed to the macro as ARG1

I added a System command to log the variables to a text file so I know  
when the page is made, the variables are correct.

[macro-pageall]
; Context for paging all devices.
;   This will search the sip table in the realtime database
;   for all phones that start with a number.  That number is
;   passed to this macro as ${ARG1}.
;
;   ARG1 = The first digit of the phones to be paged
;   ARG2 = Device for the PA system.  If the user selected to
;   page the PA system.  That will be included.
;
exten => s,1,Set(MEETME_RECORDINGFORMAT=wav)
exten => s,2,Set(MEETME_RECORDINGFILE=custom/paging/${EPOCH})
exten => s,3,System(/bin/echo "${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} $ 
{MEETME_RECORDINGFORMAT} ${MEETME_RECORDINGFILE}" >> /var/log/asterisk/ 
pagemacro_var.log)
exten => s,4,MYSQL(Connect connid ${realdb_host} ${realdb_user} $ 
{realdb_pass} ${realdb_db})
exten => s,5,MYSQL(Query resultid ${connid} SELECT\ name\ FROM\ sip\  
WHERE\ name\ LIKE\ "'${ARG1}%'")
exten => s,6,MYSQL(Fetch fetchid ${resultid} number)
exten => s,7,GoToIf($["${fetchid}" = "1"]?8:10)
exten => s,8,Set(pagedevice=${pagedevice}&SIP/${number})
exten => s,9,GoToIf($["${fetchid}" = "1"]?6:10)
exten => s,10,Set(pagedevice=${pagedevice:1})
exten => s,11,MYSQL(Clear ${resultid})
exten => s,12,MYSQL(Disconnect ${connid})
exten => s,13,GoToIf($["${ARG2}" != ""]?14:15)
exten => s,14,Set(pagedevice=${pagedevice}&${ARG2})
exten => s,15,SIPAddHeader(Call-Info:answer-after=0)
exten => s,16,SIPAddHeader(Alert-Info: Ring Answer)
exten => s,17,NoOp(Page Recording ${MEETME_RECORDINGFILE})
exten => s,18,Set(CALLERID(all)=System Page <1010>)
exten => s,19,Page(${pagedevice},r)

;On hangup, run script that will email the recording to shared  
conference.
exten => h,1,System(/var/lib/asterisk/scripts/mail_lastpage ${ARG1} $ 
{MEETME_RECORDINGFILE})
exten => h,2,Hangup()

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Re: [asterisk-users] Paging in Asterisk

[Paul Hales]

Does 'sip show peers' actually show the phone as registered?

PaulH


On Mon, 2007-10-15 at 02:05 +1000, [EMAIL PROTECTED] wrote:
> Actually, forget everything else.
> 
> Even when I simply pick up the handset and dial 6600, I get those errors 
> in console, so it's not related to paging or call files or anything 
> special, I guess..
> 
> Any ideas?
> 
> bu
> 
> [EMAIL PROTECTED] wrote:
> > Hi All,
> >
> > I've been trying to send a message to the list for the past 3 days, but 
> > I neither get bounces nor the message appearing in the list, so someone 
> > on IRC sugested I reply to an existing message.
> >
> > My subject is related to this message, although slightly different.
> >
> > Apologies if my actual messages appear in the list. Here's a paste of 
> > one of my past messages:
> > --
> > I'm playing with a PA-type setup, where people can dial a number, and 
> > Asterisk would place a call file to get another phone to dial in (auto 
> > answering) and play to it a sound.
> >
> > It's woking, but I'm getting some errors, as I'll paste below.
> >
> > So, my setup:
> > Asterisk 1.4.13
> > Debian GNU/Linux 4.0
> > Linux Kernel 2.6.18-5-686
> >
> > SIP client:
> > snom360 5.3 soft-phone
> > SIP/[EMAIL PROTECTED]
> >
> > My call file:
> > Channel: Local/[EMAIL PROTECTED]/n
> > Extension: 6600
> >
> > extensions.conf:
> >
> > [from-sip]
> > exten => 6600,1,Answer
> > exten => 6600,n,Wait(1)
> > exten => 6600,n,Playback(demo-thanks)
> > exten => 6600,n,Hangup
> >
> > [localtest]
> > exten => pa,1,SIPAddHeader(Call-Info:\;answer-after=0)
> > exten => pa,n,Dial(SIP/pa)
> >
> >
> > The Console (-rvvv):
> >
> >   -- Attempting call on Local/[EMAIL PROTECTED]/n for [EMAIL PROTECTED]:1 
> > (Retry 1)
> >   -- Executing [EMAIL PROTECTED]:1] 
> > SIPAddHeader("Local/[EMAIL PROTECTED],2", 
> > "Call-Info:;answer-after=0") in new stack
> >   -- Executing [EMAIL PROTECTED]:2] Dial("Local/[EMAIL PROTECTED],2", 
> > "SIP/pa") in new stack
> >   -- Called pa
> >   -- SIP/pa-081ddd30 is ringing
> >   -- SIP/pa-081ddd30 answered Local/[EMAIL PROTECTED],2
> > == Starting Local/[EMAIL PROTECTED],1 at default,6600,1 failed so 
> > falling back to exten 's'
> >   -- Executing [EMAIL PROTECTED]:1] Wait("Local/[EMAIL PROTECTED],1", "1") 
> > in 
> > new stack
> > [Oct 11 19:10:52] WARNING[3912]: translate.c:163 framein: no samples for 
> > gsmtolin
> > [Oct 11 19:10:52] WARNING[3912]: translate.c:163 framein: no samples for 
> > gsmtolin
> > .
> > .
> > .
> > And it keeps going, really fast, indefinately, until hung up.
> >
> > Any ideas? How about the 's' error above?
> >
> > Thanks,
> > bu
> >
> >
> > Nick Couchman wrote:
> >   
> >> Our office does not have a PA system, and in our current phone system 
> >> we have a certain extension that we dial that pages over the speaker 
> >> of all the phones in the office.  Does Asterisk support this feature? 
> >>  If so, could someone tell me the best way to set this up in AsteriskNOW?
> >>
> >>
> >> Thanks,
> >>
> >> Nick
> >>
> >> 
> >
> >
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> >   
> 
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Re: [asterisk-users] Paging in Asterisk

[bu]

Actually, forget everything else.

Even when I simply pick up the handset and dial 6600, I get those errors 
in console, so it's not related to paging or call files or anything 
special, I guess..

Any ideas?

bu

[EMAIL PROTECTED] wrote:
> Hi All,
>
> I've been trying to send a message to the list for the past 3 days, but 
> I neither get bounces nor the message appearing in the list, so someone 
> on IRC sugested I reply to an existing message.
>
> My subject is related to this message, although slightly different.
>
> Apologies if my actual messages appear in the list. Here's a paste of 
> one of my past messages:
> --
> I'm playing with a PA-type setup, where people can dial a number, and 
> Asterisk would place a call file to get another phone to dial in (auto 
> answering) and play to it a sound.
>
> It's woking, but I'm getting some errors, as I'll paste below.
>
> So, my setup:
> Asterisk 1.4.13
> Debian GNU/Linux 4.0
> Linux Kernel 2.6.18-5-686
>
> SIP client:
> snom360 5.3 soft-phone
> SIP/[EMAIL PROTECTED]
>
> My call file:
> Channel: Local/[EMAIL PROTECTED]/n
> Extension: 6600
>
> extensions.conf:
>
> [from-sip]
> exten => 6600,1,Answer
> exten => 6600,n,Wait(1)
> exten => 6600,n,Playback(demo-thanks)
> exten => 6600,n,Hangup
>
> [localtest]
> exten => pa,1,SIPAddHeader(Call-Info:\;answer-after=0)
> exten => pa,n,Dial(SIP/pa)
>
>
> The Console (-rvvv):
>
>   -- Attempting call on Local/[EMAIL PROTECTED]/n for [EMAIL PROTECTED]:1 
> (Retry 1)
>   -- Executing [EMAIL PROTECTED]:1] 
> SIPAddHeader("Local/[EMAIL PROTECTED],2", 
> "Call-Info:;answer-after=0") in new stack
>   -- Executing [EMAIL PROTECTED]:2] Dial("Local/[EMAIL PROTECTED],2", 
> "SIP/pa") in new stack
>   -- Called pa
>   -- SIP/pa-081ddd30 is ringing
>   -- SIP/pa-081ddd30 answered Local/[EMAIL PROTECTED],2
> == Starting Local/[EMAIL PROTECTED],1 at default,6600,1 failed so 
> falling back to exten 's'
>   -- Executing [EMAIL PROTECTED]:1] Wait("Local/[EMAIL PROTECTED],1", "1") in 
> new stack
> [Oct 11 19:10:52] WARNING[3912]: translate.c:163 framein: no samples for 
> gsmtolin
> [Oct 11 19:10:52] WARNING[3912]: translate.c:163 framein: no samples for 
> gsmtolin
> .
> .
> .
> And it keeps going, really fast, indefinately, until hung up.
>
> Any ideas? How about the 's' error above?
>
> Thanks,
> bu
>
>
> Nick Couchman wrote:
>   
>> Our office does not have a PA system, and in our current phone system 
>> we have a certain extension that we dial that pages over the speaker 
>> of all the phones in the office.  Does Asterisk support this feature? 
>>  If so, could someone tell me the best way to set this up in AsteriskNOW?
>>
>>
>> Thanks,
>>
>> Nick
>>
>> 
>
>
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>   


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Re: [asterisk-users] Paging in Asterisk

[bu]

Hi All,

I've been trying to send a message to the list for the past 3 days, but 
I neither get bounces nor the message appearing in the list, so someone 
on IRC sugested I reply to an existing message.

My subject is related to this message, although slightly different.

Apologies if my actual messages appear in the list. Here's a paste of 
one of my past messages:
--
I'm playing with a PA-type setup, where people can dial a number, and 
Asterisk would place a call file to get another phone to dial in (auto 
answering) and play to it a sound.

It's woking, but I'm getting some errors, as I'll paste below.

So, my setup:
Asterisk 1.4.13
Debian GNU/Linux 4.0
Linux Kernel 2.6.18-5-686

SIP client:
snom360 5.3 soft-phone
SIP/[EMAIL PROTECTED]

My call file:
Channel: Local/[EMAIL PROTECTED]/n
Extension: 6600

extensions.conf:

[from-sip]
exten => 6600,1,Answer
exten => 6600,n,Wait(1)
exten => 6600,n,Playback(demo-thanks)
exten => 6600,n,Hangup
   
[localtest]
exten => pa,1,SIPAddHeader(Call-Info:\;answer-after=0)
exten => pa,n,Dial(SIP/pa)


The Console (-rvvv):

  -- Attempting call on Local/[EMAIL PROTECTED]/n for [EMAIL PROTECTED]:1 
(Retry 1)
  -- Executing [EMAIL PROTECTED]:1] 
SIPAddHeader("Local/[EMAIL PROTECTED],2", 
"Call-Info:;answer-after=0") in new stack
  -- Executing [EMAIL PROTECTED]:2] Dial("Local/[EMAIL PROTECTED],2", 
"SIP/pa") in new stack
  -- Called pa
  -- SIP/pa-081ddd30 is ringing
  -- SIP/pa-081ddd30 answered Local/[EMAIL PROTECTED],2
== Starting Local/[EMAIL PROTECTED],1 at default,6600,1 failed so 
falling back to exten 's'
  -- Executing [EMAIL PROTECTED]:1] Wait("Local/[EMAIL PROTECTED],1", "1") in 
new stack
[Oct 11 19:10:52] WARNING[3912]: translate.c:163 framein: no samples for 
gsmtolin
[Oct 11 19:10:52] WARNING[3912]: translate.c:163 framein: no samples for 
gsmtolin
.
.
.
And it keeps going, really fast, indefinately, until hung up.

Any ideas? How about the 's' error above?

Thanks,
bu


Nick Couchman wrote:
>
> Our office does not have a PA system, and in our current phone system 
> we have a certain extension that we dial that pages over the speaker 
> of all the phones in the office.  Does Asterisk support this feature? 
>  If so, could someone tell me the best way to set this up in AsteriskNOW?
>
>
> Thanks,
>
> Nick
>


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Re: [asterisk-users] Paging in Asterisk

[Bill Andersen]

> I use a mysql script to dynamically generate the page command
> and page about 70 phones, and I have never had a reboot problem.
> Sometimes there is a slight delay waiting for all the phones to
> join the page conference. I am using a mix of 650's, 550's, and
> 330's. 
>
> It must only be an issue if you are using presence. Maybe I will
> setup presence on a couple phones and see if they reboot.

It is definitely the presence.  My set up is as follows:

IP601 (Receptionist x7110) << Buddy List turned ON
 (Only way to update sidcar)
IP501 (32 "user" phones)   << Buddy List turned OFF
 (Cuts down on traffic)

I have about 15 of the IP501s in the Page Group.  I would have
all of them, but too may more than 15 will cause the 601 to
reboot EVERY time. At least with 15, it only reboots about
1 out of 15-20 pages.  I have to have at least the 15 extensions
in my group to get the whole building covered.

When there is a page, the CLI shows

Extension Changed 7118 new state InUse for Notify User 7110
Extension Changed 7134 new state InUse for Notify User 7110
Extension Changed 7117 new state InUse for Notify User 7110
Extension Changed 7125 new state InUse for Notify User 7110
Extension Changed 7123 new state InUse for Notify User 7110
Extension Changed 7114 new state InUse for Notify User 7110
Extension Changed 7137 new state InUse for Notify User 7110
... for all 15

THIS, according to Polycom is the rush of presence information
that overwhelms the 601 and causes it to reboot.

They suggest moving to new firmware (which for right now I
can't), but I'm going to try Joseph's GREAT suggestion to
set one of the other lines on the 501 to a secondary
extension within the page group... Thanks Joseph!!!
I'll let you know how that works.

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Re: [asterisk-users] Paging in Asterisk

[Joseph Begumisa]

Yes, this is true when using presence on the 601's.  With presence disabled,
you get no reboots at all.  That's why when I realized that, I decided on
the setup I mentioned below.  However, you could let us know whether this is
true for the 650's 550's and 330's.

 

Joseph

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Forrest Beck
Sent: Friday, October 12, 2007 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Paging in Asterisk

 

I use a mysql script to dynamically generate the page command and page about
70 phones, and I have never had a reboot problem.  Sometimes there is a
slight delay waiting for all the phones to join the page conference.  I am
using a mix of 650's, 550's, and 330's.  

 

It must only be an issue if you are using presence.  Maybe I will setup
presence on a couple phones and see if they reboot.

 

 

Forrest Beck

[EMAIL PROTECTED]

http://www.shift8.biz/blog

 

 

On Oct 11, 2007, at 4:34 PM, Jim Canfield wrote:





Joseph Begumisa wrote: 

I had the same problem with 45 polycom 601 phones in the same page group.
It was just like you describe it and I got the same answer from polycom.
What I did to go around that was add a second line key with a different
extension number on each phone and then create the page group with the
second extensions as members instead of the first extension.

Interesting.  I've seen the reboot mystery mentioned before. Some have
pointed to power, but this makes sense. Do you know if this is still an
issue on the newer series (330,550,650) phones as well? 



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Re: [asterisk-users] Paging in Asterisk

[Forrest Beck]

I use a mysql script to dynamically generate the page command and  
page about 70 phones, and I have never had a reboot problem.   
Sometimes there is a slight delay waiting for all the phones to join  
the page conference.  I am using a mix of 650's, 550's, and 330's.


It must only be an issue if you are using presence.  Maybe I will  
setup presence on a couple phones and see if they reboot.



Forrest Beck
[EMAIL PROTECTED]
http://www.shift8.biz/blog


On Oct 11, 2007, at 4:34 PM, Jim Canfield wrote:


Joseph Begumisa wrote:
I had the same problem with 45 polycom 601 phones in the same page  
group.  It was just like you describe it and I got the same answer  
from polycom.  What I did to go around that was add a second line  
key with a different extension number on each phone and then  
create the page group with the second extensions as members  
instead of the first extension.


Interesting.  I've seen the reboot mystery mentioned before. Some  
have pointed to power, but this makes sense. Do you know if this is  
still an issue on the newer series (330,550,650) phones as well?



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Re: [asterisk-users] Paging in Asterisk

[Joseph Begumisa]

No, I am not sure whether it's still an issue with the newer series because
I have not had a chance to test this with one of those models.

 

Regards,

 

Joseph

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Canfield
Sent: Thursday, October 11, 2007 4:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Paging in Asterisk

 

Joseph Begumisa wrote: 

I had the same problem with 45 polycom 601 phones in the same page group.
It was just like you describe it and I got the same answer from polycom.
What I did to go around that was add a second line key with a different
extension number on each phone and then create the page group with the
second extensions as members instead of the first extension.

Interesting.  I've seen the reboot mystery mentioned before. Some have
pointed to power, but this makes sense. Do you know if this is still an
issue on the newer series (330,550,650) phones as well? 



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Re: [asterisk-users] Paging in Asterisk

[Jim Canfield]

   Joseph Begumisa wrote:

 I had the same problem with 45 polycom 601 phones in the same page
 group.  It was just like you describe it and I got the same answer from
 polycom.  What I did to go around that was add a second line key with a
 different extension number on each phone and then create the page group
 with the second extensions as members instead of the first extension.

   Interesting.  I've seen the reboot mystery mentioned before. Some have
   pointed to power, but this makes sense. Do you know if this is still an
   issue on the newer series (330,550,650) phones as well?
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