[Asterisk-Users] Polycom IP 500 Voicemail
Wiley E. Siler
wsiler at e2020inc.com
Wed Jul 21 23:21:46 MST 2004
Voicemail.conf
-------------------------------------------------------------
[general]
format=gsm
[local]
2000 => 1234,Sarah,123 at 456.com
2001 => 1234,Gene,123 at 456.com
2002 => 1234,Lee,123 at 456.com
2003 => 1234,Wiley,123 at 456.com
----------------------------------------------------------------------
Sip.conf
------------------------------------------------------------------------
---
[general]
port=5060
[2000]
type=friend
host=dynamic
context=local
allow=g711
secret=PASSWORD
callerid="Front Desk" <2000>
mailbox=2000
dtmfmode=rfc2833
nat=0
[2001]
type=friend
context=local
allow=g711
secret=PASSWORD
callerid="Gene" <2001>
mailbox=2001
dtmfmode=rfc2833
nat=0
[2002]
type=friend
host=dynamic
context=local
allow=g711
secret=PASSWORD
callerid="Lee" <2002>
mailbox=2002
dtmfmode=rfc2833
nat=0
[2003]
type=friend
host=dynamic
context=local
;allow=g729
allow=g711
secret=PASSWORD
callerid="Wiley" <2003>
mailbox=2003
dtmfmode=rfc2833
nat=0
------------------------------------------
Other Stuff
------------------------------------------
And what is this?
> [outgoing]
> exten => _9.,1,Dial(ZAP/g2/${EXTEN,1})
What is Zap/g2? I don't see group 2 given in zapata.conf.
Mistake on my part. I changed this to g1 which is correct, right?
--------------------------------------------------------------------
> Now I need to do something in oss.conf and zapata.conf to
ensure which > one answers the X100P
right?
Yeah, this is a mess. First, are we answering phone calls on
the console? If yes, you're going to
need your incoming phones to ring /dev/console. I don't
think you want this, so oss.conf can wait.
I can honestly say. I have no idea. This is where the idea of contexts
breaks aoart for me. I want to start out just making my server pass the
ring to a group of phones (see setup in original mail). Later I am
going to define some IVR stuff and have * pick up the line and route to
people on user input.
--------------------------------------------------------------
Second, why does your incoming context also include local and
outgoing?
That doesn't seem to quite right to me.
I corrected it and I will continue to try and update.
Thanks for you help!
Wiley
-----Original Message-----
From: John Baker [mailto:JohnB at listbrokers.com]
Sent: Wednesday, July 21, 2004 7:04 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail
OK, let's work on this.
> Actually, I am having trouble with my X100P setup too which will >
probably sow when you read through my configs. I cannot get my >
referencing from contaxt to context setup correctly.
First things first. I would like to see how your phones are setup in
sip.conf along with your voicemail.conf. Specifically, what context the
sip phones are put into and whether or not the extensions of the sip
phones match your voicemail boxes.
For example, from my sip.conf file for my extension 7001, I have:
[7001]
context=from-internal
callerid="John Baker" <7001>
type=friend
username=7001
secret=XXXXXXXXX
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
qualify=200 ; Qualify peer is no more than 200ms
away
protocol=udp
dtmfmode=rfc2833
mailbox=7001 at dont_spam_me
nat=0
disallow=all
allow=ulaw
allow=gsm
auth=md5
and the relevant line from voicemail.conf is
[listbrokers]
7001 => XXXXX,John Baker,JohnB at dont_spam_me.com,,tz=central
> Now I need to do something in oss.conf and zapata.conf to ensure
which > one answers the X100P right?
Yeah, this is a mess. First, are we answering phone calls on the
console? If yes, you're going to need your incoming phones to ring
/dev/console. I don't think you want this, so oss.conf can wait.
Second, why does your incoming context also include local and outgoing?
That doesn't seem to quite right to me.
And what is this?
> [outgoing]
> exten => _9.,1,Dial(ZAP/g2/${EXTEN,1})
What is Zap/g2? I don't see group 2 given in zapata.conf.
John
Wiley E. Siler wrote:
> Actually, I am having trouble with my X100P setup too which will
> probably sow when you read through my configs. I cannot get my
> referencing from contaxt to context setup correctly.
>
>
> These are in extensions.conf
> ; ----------------------------------------------
> ; GLOBALS - Defines variables for use of devices, extensions ;
> ----------------------------------------------
>
> [globals]
> ;Reception
> PHONES0=SIP/2000
> PHONES0VM=2000
>
> PHONES1=SIP/2001
> PHONES1VM=2001
>
> PHONES2=SIP/2002
> PHONES2VM=2002
>
> PHONES3=SIP/2003
> PHONES3VM=2003
>
> ;Trunk Info
> TRUNK=Zap/g1 ; Trunk interface
> TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
>
> ; ----------------------------------------------
> ; END GLOBALS
> ; ----------------------------------------------
>
>
> [macro-vmessage]
> exten => s,1,VoiceMail2(u${ARG1})
> exten => s,2,Playback(groovy)
> ;exten => s,3,BackGround(dialing)
> exten => s,3,Playback(goodbye)
> exten => s,4,Hangup
>
> ; ----------------------------------------------
> ; DEFINE EXTENSIONS
> ; ----------------------------------------------
>
> [trunkint]
> ;
> ; International long distance through trunk
> ;
> exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9011.,2,Congestion
>
> [trunkld]
> ;
> ; Long distance context accessed through trunk
> ;
> exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91NXXNXXXXXX,2,Congestion
>
> [trunklocal]
> ;
> ; Local seven-digit dialing accessed through trunk interface
> ;
> exten => _9480XXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9480XXXXXXX,2,Congestion
>
> exten => _9602XXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9602XXXXXXX,2,Congestion
>
> exten => _9623XXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9623XXXXXXX,2,Congestion
>
>
> exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9NXXXXXX,2,Congestion
>
> [trunktollfree]
> ;
> ; Long distance context accessed through trunk interface
> ;
> exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91800NXXXXXX,2,Congestion
> exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91888NXXXXXX,2,Congestion
> exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91877NXXXXXX,2,Congestion
> exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91866NXXXXXX,2,Congestion
>
> [international]
> ;
> ; Master context for international long distance
> ;
> ignorepat => 9
> include => longdistance
> include => trunkint
>
> [longdistance]
> ;
> ; Master context for long distance
> ;
> ignorepat => 9
> include => local
> include => trunkld
>
> [local]
> ;
> ; Master context for local, toll-free, and iaxtel calls only
> ;
> ignorepat => 9
> include => trunklocal
> include => trunktollfree
> ;
> ; You can use an alternative switch type as well, to resolve
> ; extensions that are not known here, for example with remote
> ; IAX switching you transparently get access to the remote
> ; Asterisk PBX
> ;
> ; switch => IAX2/user:password at bigserver/local
>
> ; now if we dial 8, we can check voicemail.
> ;
>
> ;exten => 8,1,VoicemailMain(s${CALLERIDNUM}) ;= no password
> exten => 8,1,VoicemailMain(${CALLERIDNUM}) ;= password
> exten => 8,2,Hangup
>
> ; Add some more extensions for the two extensions . now we'll be able
to
> call one extension from the other.
> ; And if no one answers, it will go to the mailbox for that extension.
> ;
> ; extension 2000
> ;
> exten => 2000,1,Dial(${PHONES1},20,trf)
> exten => 2000,2,Macro(vmessage,${PHONES0VM})
> exten => 2000,3,Hangup
> ;
> ; extension 2001
> ;
> exten => 2001,1,Dial(${PHONES1},20,trf)
> exten => 2001,2,Macro(vmessage,${PHONES1VM})
> exten => 2001,3,Hangup
> ;
> ; extension 2002
> ;
> exten => 2002,1,Dial(${PHONES2},20,trf)
> exten => 2002,2,Macro(vmessage,${PHONES2VM})
> exten => 2002,3,Hangup
> ;
> ; extension 2003
> ;
> exten => 2003,1,Dial(${PHONES3},20,trf)
> exten => 2003,2,Macro(vmessage,${PHONES3VM})
> exten => 2003,3,Hangup
>
> ;
> ; IVR CHOICE 1
> ;
> exten => 1,1,Answer
> exten => 1,2,Playback(tt-somethingwrong)
> exten => 1,3,Playback(tt-monkeysintro)
> exten => 1,4,Playback(tt-monkeys)
> exten => 1,5,Hangup
>
> ;
> ; IVR RECORDER
> ;
> ; Record voice file to /tmp directory
> exten => 205,1,Wait(2) ; Call 205 to Record new Sound Files
> exten => 205,2,Record(/tmp/asterisk-recording:gsm)
> exten => 205,3,Wait(2)
> exten => 205,4,Playback(/tmp/asterisk-recording)
> exten => 205,5,wait(2)
> exten => 205,6,Hangup
>
>
> ;
> ;MUSIC ON HOLD EXTENSION
> ;
> exten => 6000,1,Answer
> exten => 6000,2,SetMusicOnHold(default)
> exten => 6000,3,MusicOnHold()
> exten => 6000,4,Hangup
>
> ; ---------------------------------------------
> ; END DEFINE EXTENSIONS
> ; ----------------------------------------------
>
> ;------------------------------------
> ; RING EVERYONE EXTENSIONS
> ;-----------------------------------
>
> ;
> ; ring everyone
> ;
> exten =>
6001,1,Dial(${PHONES0}&${PHONES1}&${PHONES2}&${PHONES3},20,trf)
>
> exten => 6001,2,Dial(SIP/6000,20,trf)
> exten => 6001,3,Hangup
>
>
> ;-----------------------------------
> ; END RING EVERYONE
> ;----------------------------------
>
> ;----------------------------------------------
> ; DEFINE CALL PARKING AREA
> ;---------------------------------------------
> include =>parkedcalls
>
> ;--------------------------------------------------
> ; DEFINE MEETING ROOMS
> ;-------------------------------------------------
> ;exten => 4000,1,Meetme,40000
>
> exten => s,1,Answer
> exten => s,2,BackGround(greeting)
>
> exten => t,1,Playback(vm-goodbye)
> exten => t,2,HangUp
>
>
> [incoming]
> exten => s,1,Answer
> exten => s,2,Dial(SIP/2000)
> exten => s,3,Hangup
> include => local
> include => outgoing
>
> [outgoing]
> exten => _9.,1,Dial(ZAP/g2/${EXTEN,1})
>
>
>
> Now I need to do something in oss.conf and zapata.conf to ensure which
> one answers the X100P right?
>
> in zapata.conf...
>
> [channels]
>
> busydetect=1
> busycount=7
>
> relaxdtmf=yes
> callwaiting=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
>
> usecallerid=yes
>
> echocancel=yes
> echocancelwhenbridged=yes
>
> rxgain=0.5
> txgain=0.5
>
> group=1
> pickupgroup=1
>
> immediate=no
>
> signalling=fxs_ks
> callerid=asreceived
> channel=1
>
> context=incoming
>
>
> -----Original Message-----
> From: John Baker [mailto:JohnB at listbrokers.com]
> Sent: Tuesday, July 20, 2004 10:52 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail
>
> Mr. Siler -
>
> I respond in kind...
>
> > I am using the latest firmware from the Wiki. 2.4.2 I believe.
>
> Oops. The latest firmware version is 1.2.0
>
> Try http://www.freedomphones.net/polycom/files/ for the latest
firmware.
>
> If you don't show the latest version, (try pressing the right
buttons
> on your Polycom phone to get a version number) then anything else we
> discuss is worthless.
>
> > I edit my XML docs in notepad only.
>
> DON'T DO THAT!! Trust me. I wasted alot of time with a text editor.
> For a free XML editor, I use http://www.xmlcooktop.com/ Oh, and by
the
> way...
>
> USING AN XML EDITOR IS VERY IMPORTANT!!! Polycom phones will load
> corrupt XML, but not the way you want it. You will think your changes
> have an effect, but if the XML isn't good, then they won't. Test your
> settings with an XML editor!!! Make sure your config files read OK.
>
> > This retrieves my mail through the menu system but not directly.
>
> Directly to me means I press the 'Messages' button on my Polycom 600
and
> asterisk asks me for a password. (Asterisk discerns the mailbox from
> the extension of the phone) It's one touch (but still password
> protected) It's working here and I'm sure we can get it to work at
your
> office.
>
> > Voicemail answers on extension 8.
>
> Just to be sure, can I see your extensions.conf?
>
> John
>
> Wiley E. Siler wrote:
>
>>I have tried both a nul and the following...
>>
>>Subscribe = 8
>>callbackmode = contact
>>Callback = 8
>>
>>This retrieves my mail through the menu system but not directly.
>>
>>I am using the latest firmware from the Wiki. 2.4.2 I believe.
>>
>>I edit my XML docs in notepad only.
>>
>>Voicemail answers on extension 8.
>>
>>Thanks,
>>Wiley
>>
>>
>>
>>-----Original Message-----
>>From: John Baker [mailto:johnb at listbrokers.com]
>>Sent: Tuesday, July 20, 2004 3:48 PM
>>To: asterisk-users at lists.digium.com
>>Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail
>>
>>Why do you have a non-null msg.mwi.1.subscribe? You're sending a
>>SUBSCRIBE request to asterisk at extension '8' upon bootup. Is that
>>what you want?
>>
>>Did you upgrade the phone with the latest firmware?
>>
>>Did you use an XML editor to mess with the configuration? I messed up
>
>
>>mine once using a text editor.
>>
>>Is asterisk setup to answer voicemail at extension '8'?
>>
>>Try the above and let me know.
>>
>>John
>>
>>On Tue, 2004-07-20 at 11:52, Wiley E. Siler wrote:
>>
>>
>>>I tried this configuration and it still does not work for me. In
>>>fact, now I cannot dial in using the menu system of the message
>>>center. Here is how I have now mine configured and what I get...
>>>
>>><msg msg.bypassInstantMessage="1">
>>> <mwi msg.mwi.1.subscribe="8"
>>>msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="8"
>>>msg.mwi.2.subscribe="" msg.mwi.2.callBackMode="registration"
>>>msg.mwi.2.callBack="" msg.mwi.3.subscribe=""
>>>msg.mwi.3.callBackMode="registration" msg.mwi.3.callBack=""
>>>msg.mwi.4.subscribe="" msg.mwi.4.callBackMode="registration"
>>>msg.mwi.4.callBack="" msg.mwi.5.subscribe=""
>>>msg.mwi.5.callBackMode="registration" msg.mwi.5.callBack=""
>>>msg.mwi.6.subscribe="" msg.mwi.6.callBackMode="registration"
>>>msg.mwi.6.callBack=""/>
>>> </msg>
>>> <nat nat.ip="" nat.signalPort="" nat.mediaPortStart=""/>
>>> <user_preferences up.headsetMode="0" up.useDirectoryNames="0"
>>>up.oneTouchVoiceMail="1"/>
>>>
>>>
>>>
>>>The relevent fields being the msg. fields and up.oneTouchVoicemail
>>>
>>>This allows me voicemail via the Messages button but it is not
direct.
>>>I have to navigate still through allt he menus.
>>>
>>>W
>>>
>>>
>>>
>>>-----Original Message-----
>>>From: John Baker [mailto:JohnB at listbrokers.com]
>>>Sent: Monday, July 19, 2004 10:17 PM
>>>To: asterisk-users at lists.digium.com
>>>Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail
>>>
>>>My Polycom Message button goes straight to voicemail. Here's how:
>>>
>>>1) Use the latest firmware, available on the Wiki
>>>
>>>2) In your phone.cfg file (for each phone) set
>>>
>>><msg msg.bypassInstantMessage="1">
>>><mwi msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact"
>>>msg.mwi.1.callBack="76" .... >
>>>
>>>3) In your extensions.conf, have something like:
>>>
>>>exten => 76,1,VoiceMailMain2(${EXTEN}@whatever_you_have_here)
>>>
>>>(Let's assume your voice mailbox is the same as your extension)
>>>
>>>Then when you push the message button, asterisk will ask for your
>>>password! You're in!
>>>
>>>John
>>>
>>>
>>>Chris A. Icide wrote:
>>>
>>>
>>>>On 04:28 PM 7/19/2004, Wiley E. Siler wrote:
>>>>
>>>>>Mine does the same. Once in Message center I can choose selection
>>>>>1.Message Center and then soft key Select. Then I select the
>>>>>registered line that I want to check voice mail on. That is no
>>>>
>>>>less than
>>>>
>>>>>4 key strokes just to get into your voice mail. Not many to me
>>>>
>>>>but tons >to an unskilled user. However, in the documentation
>>>>regarding the >bypassInstantMessage value, supposedly, setting
>>>>bypassInstantMessage to
>>>>
>>>>>1 is supposed to allow you to go right into voice mail without
>>>>
>>>>>navigating the Message Center. That is the big question on my mind
>>>>
>>>>at >this point. I have yet to get this to work and I also don't
>>>>think I am >receiving any SIMPLE messages ti show me that I have
>>>
>>>messages waiting.
>>>
>>>
>>>>>Do you get a message waiting indicator?
>>>>>
>>>>
>>>>I do get MWI, there are a few things you need to set, and I forget
>>>>what off the top of my head, soon as I can look and post it here.
>>>>
>>>>I haven't tried the bypassInstantMessage value, but I'll take a look
>>
>>
>>>>and see if I can get it to work.
>>>>
>>>>-Chris
>>>>
>>>>_______________________________________________
>>>>Asterisk-Users mailing list
>>>>Asterisk-Users at lists.digium.com
>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>To UNSUBSCRIBE or update options visit:
>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>>
>>>
>>>_______________________________________________
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>>>
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>>>
>>
>>
>>
>>_______________________________________________
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>>
>>
>
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