[asterisk-biz] Successful Switchvox Deployment

Tue Oct 13 17:35:59 CDT 2009

Not really...

Here's my approach it verifies if user on phone, and wont nagg them

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

[intercom]

exten => s,1,Answer
exten => s,2,Playback(beep)
exten => s,3,Set(TIMEOUT(digit)=5)
exten => s,4,WaitExten(10)

exten => 777,1,SIPAddHeader(Call-Info: <sip:192.168.16.105>\;answer-after=0)
exten => 777,2,Page(Local/306 at page&Local/303 at page&Local/998 at page)

[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,SIPAddHeader(Call-Info: <sip:192.168.1.100>\;answer-after=0) ;
asterisk IP
exten => s,4,Dial(${ARG1}|3|) ; should ring 3 seconds
exten => s,5,Hangup
exten => s,105,Hangup 

[page] ; Paging context
exten => _X.,1,Macro(page,SIP/${EXTEN})

>>-----Original Message-----
>>From: asterisk-biz-bounces at lists.digium.com [mailto:asterisk-biz-
>>bounces at lists.digium.com] On Behalf Of Jimmy Godbout
>>Sent: October-13-09 10:14 AM
>>To: Commercial and Business-Oriented Asterisk Discussion
>>Subject: Re: [asterisk-biz] Successful Switchvox Deployment
>>
>>Hi,
>>I got the same problem about a year and a half ago. If you have call-
>>waiting enable, it won't matter because it will put the actual call on-
>>hold and play the message. You have to either turn it off or you have
>>to remove the phone(s) that are in use from the list of the intercom.
>>Since you might not have access to the dialplan with this distribution,
>>you might go with the first option and turn the call-waiting feature
>>off.
>>
>>Jimmy
>>
>>> -----Original Message-----
>>> From: lists at contacttel.com
>>> Sent: Sun, 11 Oct 2009 22:32:16 -0400
>>> To: asterisk-biz at lists.digium.com
>>> Subject: Re: [asterisk-biz] Successful Switchvox Deployment
>>>
>>> Same. I think we need yes.
>>>
>>>
>>>
>>> That way it will sned trough speaker lol and not the first line..
>>which
>>> puts
>>> the customer on phone already on hold
>>>
>>>
>>>
>>> From: asterisk-biz-bounces at lists.digium.com
>>> [mailto:asterisk-biz-bounces at lists.digium.com] On Behalf Of Brian
>>> Franklin
>>> Sent: October-11-09 10:33 PM
>>> To: Commercial and Business-Oriented Asterisk Discussion
>>> Subject: Re: [asterisk-biz] Successful Switchvox Deployment
>>>
>>>
>>>
>>> Its set to 'no', how about yours?
>>>
>>>
>>>
>>> Brian
>>>
>>> From: asterisk-biz-bounces at lists.digium.com
>>> [mailto:asterisk-biz-bounces at lists.digium.com] On Behalf Of
>>> lists at contacttel.com
>>> Sent: Sunday, October 11, 2009 6:11 PM
>>> To: 'Commercial and Business-Oriented Asterisk Discussion'
>>> Subject: Re: [asterisk-biz] Successful Switchvox Deployment
>>>
>>>
>>>
>>> "intercom paging via phones does not work properly with
>>Switchvox/Linksys
>>> phones, puts existing calls on hold and seems to be the cause an
>>issue
>>> that
>>> requires a PBX reboot the next day"
>>>
>>>
>>>
>>> Got same problem here. is your send audio to speaker set to yes or no
>>?
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> From: asterisk-biz-bounces at lists.digium.com
>>> [mailto:asterisk-biz-bounces at lists.digium.com] On Behalf Of Brian
>>> Franklin
>>> Sent: October-11-09 6:47 PM
>>> To: Commercial and Business-Oriented Asterisk Discussion
>>> Subject: [asterisk-biz] Successful Switchvox Deployment
>>>
>>>
>>>
>>> I just wanted to share the details for a recent successful VoIP
>>> deployment:
>>>
>>>
>>>
>>> Project:
>>>
>>> Replace Mitel SX-200 TDM/VoIP hybrid PBX for 90 user environment.
>>> Integrate
>>> with existing Fax server, PA system, credit card and adhoc fax
>>machines
>>>
>>>
>>>
>>> Solution:
>>>
>>> 1 Adtran Atlas 550 - allows existing PRI's to be shared between PBX
>>and
>>> Fax
>>> server
>>>
>>> 2 Adtran 1238 48 port PoE switches - so far, this is a great switch
>>>
>>> 1 Switchvox SMB server - largest appliance available
>>>
>>> 4 Cisco SPA509G - call center users
>>>
>>> 20 Cisco SPA525G - senior management
>>>
>>> 65+ Cisco SPA942 - general users
>>>
>>> 1 Grandstream FXS gateway - credit card and adhoc fax machines
>>>
>>> 1 TFTP server - free Solarwinds TFTP server running on existing
>>windows
>>> server
>>>
>>> 1 Cyberdata VoIP paging gateway - connects VoIP to existing paging
>>system
>>>
>>>
>>>
>>> Lessons Learned
>>>
>>> -          Grandstream is not ready for primetime, requires regular
>>> reboots,
>>> will be replacing it with an Audiocodes 24 FXS gateway ASAP
>>>
>>> -          Cisco SAP525G has all the bell and whistles, but are
>>> unreliable,
>>> 3 have failed in the first week, also required me to disable  ALAW
>>and
>>> ULAW
>>> to fix buzzing sound, will probably replace with SPA962
>>>
>>> -          Intercom paging via phones does not work properly with
>>> Switchvox/Linksys phones, puts existing calls on hold and seems to be
>>the
>>> cause an issue that requires a PBX reboot the next day
>>>
>>> -          Updated Cyberdata with wrong firmware (my mistake), unit
>>would
>>> not reboot and requires it to be re-flashed by manufacturer (poor
>>> design), I
>>> had a new one shipped and left the firmware alone
>>>
>>> -          Linksys phones uses individual config files for each
>>phone,
>>> this
>>> is a problem when you have 90 phones and need to make a change to all
>>> phones, found a nice little utility that can find/replace text within
>>> multiple files, this was a huge time saver
>>>
>>>
>>>
>>> Hope this helps and/or encourages someone J
>>>
>>>
>>>
>>> Brian Franklin
>>>
>>> www.ntginc.net
>>
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