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asterisk and MWI

HI All,

We are running a digium aa50 with grandstream gxp 2020.

I am unable to work out how to get the MWI light showing when we have new messages on the system.

We have enabled "Subscribe for MWI" and disabled "subscribe for registration event" and "Publish for presence".

Any help would be appreciated.
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Cheryl Lander
Asked:
Cheryl Lander
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1 Solution
 
feptiasCommented:
Asterisk doesn't usually require explicit MWI subscription, so it may be better to disable "Subscribe for MWI". It should automatically get told about messages provided Asterisk knows which voicemail box is associated with each phone. You can check that using the CLI command:
  sip show peer <extn_number>

In the list of data that is displayed, look for the field "Mailbox". It should contain the voicemail box number.

The Grandstream field "Voicemail User ID" is misleading - it is not the voicemail box number, but is the number the phone calls to access messages (well, that is true on the GXP2000 anyway).

Check my article for more details:
http://kb.smartvox.co.uk/index.php/asterisk/how-it-works/sip-subscribenotify-asterisk-hints-explained/
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Cheryl LanderAuthor Commented:
Ive turned off mwi subscription and rebooted the phone.

ive then gone into cli and got the following results.

Also the Voicemail User ID is of the number to access messages.

So in this case.

User # = 6001.
Number to access messages = 6050


* Name       : 6001
  Secret       : 
  MD5Secret    : 
  Context      : DLPN_DialPlan1
  Subscr.Cont. : 
  Language     : 
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 
  Pickupgroup  : 
  Mailbox      : 6001
  VM Extension : 6050
  LastMsgsSent : 1/5
  Call limit   : 100
  Dynamic      : Yes
  Callerid     : "Joe Bloggs" <6001>
  MaxCallBR    : 384 kbps
  Expire       : 3385
  Insecure     : no
  Nat          : RFC3581
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: No
  Overlap dial : Yes
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       : 
  Addr->IP     : 192.168.1.4 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 6001
  SIP Options  : (none)
  Codecs       : 0x180e (gsm|ulaw|alaw|g726|g722)
  Codec Order  : (alaw:20,ulaw:20,gsm:20,g726:20,g722:0)
  Auto-Framing:  No 
  Status       : Unmonitored
  Useragent    : Grandstream GXP2020 1.2.3.3
  Reg. Contact : sip:6001@192.168.1.4:5060;transport=udp

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feptiasCommented:
Everything looks correct. If someone leaves a message in voicemail box 6001, you should definitely see an indication on the phone. ...but you're saying that you don't. That is very odd. Can you capture the SIP packets and post them here please as follows:
1. Clear all messages from box 6001
2. At the CLI, type "sip set debug peer 6001"
3. Leave a message in box 6001
4. Save the CLI output to a file and post here as an attachment.

I use Putty to reach the CLI in Asterisk. It has the ability to buffer hundreds of lines of output and other options to allow capture to file of a large stream of output as descibed above. There are other ways of capturing the SIP. For example, by adding verbose to the options for the log file in /etc/asterisk/logger.conf, then do logger reload, then run the test above. That will write the SIP to the /var/log/asterisk/messages file.

By the way, have you checked for updates to the firmware on the phone? I am very surprised this is not just working.
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Cheryl LanderAuthor Commented:
No Good.

Will call it day.

Too much time spent on this phone ;-)

Thanks.
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