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An analog phone on a VG204 is not rolling over to voicemail. What is needed to get the calls to roll over to voicemail if the call is unanswered?

amanchau
Level 1
Level 1

Here's the config for the port and extension. Please let me know if you need any other info.
voice-port 0/0
no battery-reversal
station-id name Fuse Cashier
station-id number 20372
!
dial-peer voice 20372 pots
description Fuse Cashier
destination-pattern 20372
port 0/0
!
Thanks.

1 Accepted Solution

Accepted Solutions

I myself have never found much value in configuring these devices as SIP, and prefer SCCP or MGCP for management. The VG204 is a smaller port model so dial peers and the like are more manageable.  I don't think it wouldn't work just as your finding out maybe a little more difficult to send. I would have that gateway configured SCCP or MGCP in the time it would take me to research how to do it via sip.  

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4 Replies 4

Gregory Brunn
Spotlight
Spotlight

Your not using MGCP or SCCP to control this port from CUCM? Your sending calls to via SIP or H.323?

 

For an easy set up I would use MGCP or SCCP and just configure on the Directory Number level in CUCM.  This is of course assuming you have CUCM, but if you are voicemail my assumption is Unity Connection and not CUE.

 

Can you talk about your topology a little more?

 

 

Hi Greg,

 

Thanks, yes we have a sip trunk from CUCM to this GW. So this GW is under the destination address of the SIP Trunk on which this port is and on which we made this configuration.

 

Also yes we do have cisco unity connection in the setup.

 

So you reckon this might not work with SIP ?

 

Thanks

 

Aman

I myself have never found much value in configuring these devices as SIP, and prefer SCCP or MGCP for management. The VG204 is a smaller port model so dial peers and the like are more manageable.  I don't think it wouldn't work just as your finding out maybe a little more difficult to send. I would have that gateway configured SCCP or MGCP in the time it would take me to research how to do it via sip.  

Spoiler
 

That fixed it greg, thanks