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Wed 01 of Aug, 2007 [09:30 UTC]

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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
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Asterisk Grandstream Paging

Here is an example of two paging methods that can be used from the Grandsteam GXP-2000 and the GXP-2020 phones.

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 2020) 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 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

[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})


[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
Created by Bryant Zimmerman, Last modification by Bryant Zimmerman on Wed 11 of Jul, 2007 [21:27 UTC]

Comments Filter

* is not a good choice

by victor on Thursday 12 of July, 2007 [14:21:20 UTC]
The * prefix is the default for dialing a Voice Mail direct.

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