Valcom PagePro VIP204 Paging Gateway Overview

The PagePro IP Model VIP-204  is a SIP based paging server designed for use with virtually all IP based telephone systems. The unit provides 8 zones, groups thereof and all call paging. Up to 4 simultaneous non conflicting pages.  This PagePro IP unit can drive up to 65,000 IP speakers or 65,000 analog speakers or provide the audio source for legacy paging systems. It can simultaneously drive any combination of these paging applications. The VIP-204 can form the foundation of most paging projects. The 8 zones can be accessed individually or in flexible groups.  Multiple VIP-204 devices can be grouped together for expandability.  Provides feedback elimination.

Valcom PagePro IP VIP-204 allows paging from any SIP compliant telephone to 8 SIP page destinations. The SIP page destinations can be any of the VIP-204’s four analog outputs, any combination of those outputs, or any
combination of page groups from other Valcom VIP units. The four analog outputs on the VIP-204 are line level –10dBm audio outputs. These outputs are designed to directly drive up to 150 Valcom self-amplified speakers.
The outputs are also suitable for connection to an auxiliary input on a utility amplifier of a legacy paging system.

Group Paging Features
The PagePro IP Model VIP-204 allows SIP based phone systems to page any number of Valcom IP speakers, Valcom Amplified Speakers and Valcom VIP series analog paging gateways. Valcom multicast group paging features robust grouping and priority controls. These controls make it easy to distribute low priority announcements yet allow high priority paging to override ongoing announcements. The PagePro IP Model VIP-204 provides 8 IP zones and 1 analog output capable of driving a mix of Valcom self amplified overhead paging speakers and horns. You can assign one SIP access phone number for the analog port. The VIP-204 provides 8 fully customizable SIP identities that can access any combination of IP zones and the analog port. Analog paging functions can be seamlessly combined IP paging functions allowing powerful combinations of analog and IP paging.Group Paging Priority Feature
Any page group or port can be assigned a priority. These priorities allow a logical scaled approach to group paging. An absolute priority can be assigned to allow for emergency override in any circumstance. Group paging priorities work locally or across ganged units in the same fashion allowing facility wide emergency paging. Red Phone functionality is easily achieved with either IP phone or analog handsets via a Valcom gateway.Feedback Elimination Feature
The PagePro IP Model VIP-204 provides programmable store and forward capability for all paging accesses. This features records up to 30 seconds of audio, and then replays the audio when the page call is terminated. The number of times the recorded audio is played back can be set from 1 to 5. Some paging applications require live announcements. For these applications the store and forward capability can be turned on/off per page zone.Wiring Supervision Features
The PagePro IP Model VIP-204 provides for supervision of field wiring. If a fault is detected in the wiring connecting the analog output to either speakers or amplifiers it can be reported to an external device or syslog facility.Audio Supervision Features
The PagePro IP Model VIP-204 provides internal monitoring of the audio path from the network connection to the analog output connector and can report failures to an external device or syslog facility.Legacy Applications
The PagePro IP Model VIP-204 provides 1 analog output that can easily be conditioned to drive any existing overhead paging system. Legacy paging features can be seamlessly combined allowing powerful combinations of legacy analog and IP paging.Easy To InstallLive or Recorded Messages65,000 Speakers, Expandable To 256 IP Paging ZonesEasily Connect Loudspeaker System To Existing Computer NetworksCompatible With Cisco, Nortel, Avaya, Iwatsu And Most IP Telephone SystemsSupports IP And Analog Amplified SpeakersSupports IP Talkback SpeakersSupports Legacy Paging SystemsSynchronized Clock CompatibilityPoE (802.3af) Or Local PowerSIP And MulticastNetwork SupervisionFeedback EliminatorNight RingDHCP

Valcom PagePro VIP204 Paging Gateway Overview

The PagePro IP Model VIP-204  is a SIP based paging server designed for use with virtually all IP based telephone systems. The unit provides 8 zones, groups thereof and all call paging. Up to 4 simultaneous non conflicting pages.  This PagePro IP unit can drive up to 65,000 IP speakers or 65,000 analog speakers or provide the audio source for legacy paging systems. It can simultaneously drive any combination of these paging applications. The VIP-204 can form the foundation of most paging projects. The 8 zones can be accessed individually or in flexible groups.  Multiple VIP-204 devices can be grouped together for expandability.  Provides feedback elimination.

Valcom PagePro IP VIP-204 allows paging from any SIP compliant telephone to 8 SIP page destinations. The SIP page destinations can be any of the VIP-204’s four analog outputs, any combination of those outputs, or any
combination of page groups from other Valcom VIP units. The four analog outputs on the VIP-204 are line level –10dBm audio outputs. These outputs are designed to directly drive up to 150 Valcom self-amplified speakers.
The outputs are also suitable for connection to an auxiliary input on a utility amplifier of a legacy paging system.

Group Paging Features
The PagePro IP Model VIP-204 allows SIP based phone systems to page any number of Valcom IP speakers, Valcom Amplified Speakers and Valcom VIP series analog paging gateways. Valcom multicast group paging features robust grouping and priority controls. These controls make it easy to distribute low priority announcements yet allow high priority paging to override ongoing announcements. The PagePro IP Model VIP-204 provides 8 IP zones and 1 analog output capable of driving a mix of Valcom self amplified overhead paging speakers and horns. You can assign one SIP access phone number for the analog port. The VIP-204 provides 8 fully customizable SIP identities that can access any combination of IP zones and the analog port. Analog paging functions can be seamlessly combined IP paging functions allowing powerful combinations of analog and IP paging.Group Paging Priority Feature
Any page group or port can be assigned a priority. These priorities allow a logical scaled approach to group paging. An absolute priority can be assigned to allow for emergency override in any circumstance. Group paging priorities work locally or across ganged units in the same fashion allowing facility wide emergency paging. Red Phone functionality is easily achieved with either IP phone or analog handsets via a Valcom gateway.Feedback Elimination Feature
The PagePro IP Model VIP-204 provides programmable store and forward capability for all paging accesses. This features records up to 30 seconds of audio, and then replays the audio when the page call is terminated. The number of times the recorded audio is played back can be set from 1 to 5. Some paging applications require live announcements. For these applications the store and forward capability can be turned on/off per page zone.Wiring Supervision Features
The PagePro IP Model VIP-204 provides for supervision of field wiring. If a fault is detected in the wiring connecting the analog output to either speakers or amplifiers it can be reported to an external device or syslog facility.Audio Supervision Features
The PagePro IP Model VIP-204 provides internal monitoring of the audio path from the network connection to the analog output connector and can report failures to an external device or syslog facility.Legacy Applications
The PagePro IP Model VIP-204 provides 1 analog output that can easily be conditioned to drive any existing overhead paging system. Legacy paging features can be seamlessly combined allowing powerful combinations of legacy analog and IP paging.Easy To InstallLive or Recorded Messages65,000 Speakers, Expandable To 256 IP Paging ZonesEasily Connect Loudspeaker System To Existing Computer NetworksCompatible With Cisco, Nortel, Avaya, Iwatsu And Most IP Telephone SystemsSupports IP And Analog Amplified SpeakersSupports IP Talkback SpeakersSupports Legacy Paging SystemsSynchronized Clock CompatibilityPoE (802.3af) Or Local PowerSIP And MulticastNetwork SupervisionFeedback EliminatorNight RingDHCP

Valcom PagePro VIP204 Paging Gateway Overview

The PagePro IP Model VIP-204  is a SIP based paging server designed for use with virtually all IP based telephone systems. The unit provides 8 zones, groups thereof and all call paging. Up to 4 simultaneous non conflicting pages.  This PagePro IP unit can drive up to 65,000 IP speakers or 65,000 analog speakers or provide the audio source for legacy paging systems. It can simultaneously drive any combination of these paging applications. The VIP-204 can form the foundation of most paging projects. The 8 zones can be accessed individually or in flexible groups.  Multiple VIP-204 devices can be grouped together for expandability.  Provides feedback elimination.

Valcom PagePro IP VIP-204 allows paging from any SIP compliant telephone to 8 SIP page destinations. The SIP page destinations can be any of the VIP-204’s four analog outputs, any combination of those outputs, or any
combination of page groups from other Valcom VIP units. The four analog outputs on the VIP-204 are line level –10dBm audio outputs. These outputs are designed to directly drive up to 150 Valcom self-amplified speakers.
The outputs are also suitable for connection to an auxiliary input on a utility amplifier of a legacy paging system.

Group Paging Features
The PagePro IP Model VIP-204 allows SIP based phone systems to page any number of Valcom IP speakers, Valcom Amplified Speakers and Valcom VIP series analog paging gateways. Valcom multicast group paging features robust grouping and priority controls. These controls make it easy to distribute low priority announcements yet allow high priority paging to override ongoing announcements. The PagePro IP Model VIP-204 provides 8 IP zones and 1 analog output capable of driving a mix of Valcom self amplified overhead paging speakers and horns. You can assign one SIP access phone number for the analog port. The VIP-204 provides 8 fully customizable SIP identities that can access any combination of IP zones and the analog port. Analog paging functions can be seamlessly combined IP paging functions allowing powerful combinations of analog and IP paging.Group Paging Priority Feature
Any page group or port can be assigned a priority. These priorities allow a logical scaled approach to group paging. An absolute priority can be assigned to allow for emergency override in any circumstance. Group paging priorities work locally or across ganged units in the same fashion allowing facility wide emergency paging. Red Phone functionality is easily achieved with either IP phone or analog handsets via a Valcom gateway.Feedback Elimination Feature
The PagePro IP Model VIP-204 provides programmable store and forward capability for all paging accesses. This features records up to 30 seconds of audio, and then replays the audio when the page call is terminated. The number of times the recorded audio is played back can be set from 1 to 5. Some paging applications require live announcements. For these applications the store and forward capability can be turned on/off per page zone.Wiring Supervision Features
The PagePro IP Model VIP-204 provides for supervision of field wiring. If a fault is detected in the wiring connecting the analog output to either speakers or amplifiers it can be reported to an external device or syslog facility.Audio Supervision Features
The PagePro IP Model VIP-204 provides internal monitoring of the audio path from the network connection to the analog output connector and can report failures to an external device or syslog facility.Legacy Applications
The PagePro IP Model VIP-204 provides 1 analog output that can easily be conditioned to drive any existing overhead paging system. Legacy paging features can be seamlessly combined allowing powerful combinations of legacy analog and IP paging.Easy To InstallLive or Recorded Messages65,000 Speakers, Expandable To 256 IP Paging ZonesEasily Connect Loudspeaker System To Existing Computer NetworksCompatible With Cisco, Nortel, Avaya, Iwatsu And Most IP Telephone SystemsSupports IP And Analog Amplified SpeakersSupports IP Talkback SpeakersSupports Legacy Paging SystemsSynchronized Clock CompatibilityPoE (802.3af) Or Local PowerSIP And MulticastNetwork SupervisionFeedback EliminatorNight RingDHCP

Asterisk Paging and Intercom

On legacy phone systems you can find the following kinds of paging: Dial a code to connect to a separate overhead paging and announcement system (like in an airport) Dial a code and connect directly to a built-in one-way announcement speaker on one or more phones Dial a code and connect directly to a built-in two-way announcement and talkback function on one or more phones
Some overhead paging systems also provide a talkback system so that the person being paged can just speak to respond. Background noise issues limit where this feature can be used. The talkback function is usually setup to be hands free. That means that the person responding to the page does not need to take any action other then speaking.

New in Asterisk 1.2: The new dialplan command Page utilizes MeetMe to page one or more phones.

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

 MulticastRTP///

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

There is also another system available since many years, the best one, combining paging and intercom. Here the talback system is limited to only one phone. The paging is done in one way mode through a group of phones, and the person being paged can respond pressing a digit to switch the nearer phone to two-way mode, simultaneously hanging-up all other phones speakers.
This mode combine the best of the two world, eliminate the noise problems, and keep the communication private as soon as the paged person pressed the right digit on a phone.
It should be possible to implement this mode on Asterisk with a managed conference and a feature map application.

Multicasting begin to be supported at all major phones manufacturers, Aastra firmware v2.2, Snom v7, Linksys,... allow the setting of a multicast listening address. This will permit to reduce the generated trafic for an extensive paging.

If a phone is in use when a page arrives, some systems can do a "whisper page" so that only the person being paged can hear the page.

SIP phones for the most part don't support any of these phone based paging functions. If a SIP phone offers an Auto Answer function, you can approximate limited paging intercom functionality. The phones most often mentioned supporting this are:

There is an 'allpage.agi' now available at http://aussievoip.com.au/allpage.agi. Documentation is available in the file. This should work with Snom and Grandstream GXP2000 phones (and possibly budgettones if they roll the changes across) with firmware greater than 1.0.13 (not publically available at time of writing, due out in October 2005)

Intercom DOES work with the Snom 200 as the mailing list link above shows. Tested on 12/20/04 with firmware 2.04g on Snom 200. One change for that posting is that the variable called in the dialplan must read "_VXML_URL" instead of "VXML_URL". Howeverr, the 'correct' way of doing Paging/Intercom is with SipAddHeader. See allpage.agi for example code.

Some analog phones have an Auto Answer function. These phone are often used in door phone systems.

ADSI phones can be configured to Auto Answer if sent the right set of signals. For information on how to do this, contact http://www.sayson.com.

Some older analog answering machines have a remote intercom function that can be used for overhead paging. Examples:

For overhead paging, you can make an Asterisk Extension go to the sound card, and wire its output to a traditional external paging system. You can also get boxes to interface an phone FXO or FXS port directly to a sound system. Examples:

Another possiblity for overhead paging is using hallway speakers that have a direct VOIP connection. Examples:


Another way to automate this is with Backticks. Someone has posted a method of using Backticks and shell scripts to dial all phones automatically. Another method for overhead paging is to solder a cable, with an RCA jack(or whatever you need), directly to the speaker of a phone that provides auto-answer. This cable can be connected directly to your amp or sound sytem used for paging.

Setting up paging with a sound card

You can use the Grandstream Budgetone phone mentioned above, it even has a round punch-out that can be used to run your cable through. Using the Grandstream as interface to the paging system is a low-cost solution that has a proven track record. With a total investment of $80 for the phone, wire, and connectors you can have a basic paging system at your office. A second unit at a remote office or warehouse makes it easy to have paging across the street, or on the other side of the world.
open up the phone and splice a connector jack in place of the builtin speaker. You can use a female RCA jack or a mini-stereo jack. jack can easily be mounted in the side of the case and used to connect to a traditional paging amplifier or amplified computer speakers. the reboot process as outlined on Asterisk phone grandstream budgetone works quite well for keeping these phones registered on the Asterisk. We've set them to reboot every four hours and have enjoyed over six months without a single user complaint. The Grandstream GXP-2000 would also work well for this- it has a 3.5mm audio jack built in. I have also read that the new redesigned BT100 series also has a headset jack.
The Grandstream GXP-2000 works very well for overhead paging. You can punch down on a 66 block a 3.5mm jack cable which then connects to your paging system. With the four sip accounts you can customize paging for different departments by having a different ring tone configured. I have this connected to an older Valcom 9970 Single Zone unit and two Handytone 488 attached to two 9 Zone Valcom 1109RTVAs. The 1109RTVA unit accepts your dtmf 0-9 (0 all call) to determine which zone to page. I can now page across the VPN to other buildings. Make sure you set the HT-488 FXO Port PSTN Silence Timeout to 10 seconds instead of 60 for paging. This reduces lockups. Also change FAX mode from t38 to Pass Thru. This is firmware 1.0.3.44 bootloader 1.0.8.11 - diver We have worked with Grandstream to develop a dial plan example that lets you use both the built-in paging function as well as a dedicated prefix method for intercom Asterisk Intercom/Paging with Grandsteam (Revised for GXP-21xx Series Phones) 10-22-2010 - BEZ(zktech)
Asterisk | Asterisk Configuration | Channel Configuration | Configuration for Specific Phones

Asterisk Grandstream Paging

(Revised for GXP-21xx Series Phones) 10-22-2010

Here is an example of two paging methods that can be used from the Grandsteam GXP-20XX and GXP-21XX series.

These methods can be extended to do many paging tasks. Our favorite is using the GXP-XXXX intercom button to trigger a meetme session with all the extensions in an office.

The first method uses the Paging function in the phone.
This is the most natural and allows you to use the phones built-in feature.
You select your line and hit the OK (GXP-2000) button or the MENU (GXP-20XX,GXP-21XX) button
This puts the phone into paging mode then you dial the extension.

The second method is to prefix the extension with an * character.
This is was the traditional method offered by Grandstream on their site, I have provided them with the first method and they are working on a white paper using the first method.

This example was written to run on Asterisk 1.4.x
(We have run modified version with 1.6.x and 1.8.x as well, on 1.8 you must modify the ChanIsAvail it appears that jumping has been dropped.)
Will post a more up to date 1.8.x version when we get ready to go production. - BEZ (zktech)

[Ext-IN]
exten => _2XXX,n,Set(l_Exten=${EXTEN})
exten => _2XXX,n,Set(l_IsPaging=${SIP_HEADER(Call-Info)})
exten => _2XXX,n,GotoIf($["${l_IsPaging}"="answer-after=0"]?callInterCom|1)
exten => _2XXX,n,Goto(Internal,${EXTEN},1)

exten => _*2XXX,1,Set(l_Exten=${EXTEN})
exten => _*2XXX,n,Goto(callInterCom,1)

exten => callInterCom,1,Macro(CoreExtPage,SIP/${l_Exten})

Please note: The s2 line has been modifed for full RFC compliance: This is requried for GXP-21XX series phones. It is backwards compatiable to the GXP-20XX series phones.

[macro-CoreExtPage]
exten => s,1,ChanIsAvail(${ARG1}|js) ; j is for dump and s is for ANY call
exten => s,2,SIPAddHeader(Call-Info: \;answer-after=0)
exten => s,3,NoOp() ; Add others here
exten => s,4,Dial(${ARG1}|j)
exten => s,5, Hangup
exten => s,105,Hangup

- BEZ (zktech) http://www.zktech.com

Please note this Grandstream link is based on the old version of our examples but may be of some help.
Grandstream Paging PDF