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Can I club two Cisco UC 520

Hi I have a senario where we have one UC 520 ( PRI  Intreface)with 40 users, now number of users has grown to 70, can I connect one more UC 520 to club all users on same PRI, we dont require any more telco line.  

Pls. reply asap
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vijaydawda
Asked:
vijaydawda
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1 Solution
 
bkepfordCommented:
Yes you can you just setup voip dial peers to trunk the call to the other UC. The problem is that you need to segregate your network into two seperate subnets or voice VLANS as each set of IP phones need to get their TFTP server info to point them to the correct UC520. Lastly the make sure you redo all of the default dial plan numbering schemes so your UCs don't overlap it makes for easier deployment.
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vijaydawdaAuthor Commented:
how can can i configure i voip dial peers to trunk the call to the other UC
do you have any sample config.
<input id="gwProxy" type="hidden"><!--Session data--><input onclick="jsCall();" id="jsProxy" type="hidden">
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bkepfordCommented:
Ok what the 71000 dial peer does is send outgoing PSTN calls out the 71 is actually stripped off at the other end and then matches the 9,XXX-XXXX dial pattern. The 72000 is the same for the other side so that they can dial out using their PSTN. This is for a small office with little UCs (not very big processors) so I made the choice to use the G711 codec with no VAD as they have plenty of bandwidth.

The Translation pattern is for removing the 72 after it arrives.

This last two dial peers are to send internal calls to the correct UC.
voice translation-rule 72
 rule 1 /^72\(........\)$/ /\1/
!
voice translation-profile SITEA
 translate called 72
!
dial-peer voice 71000 voip
 destination-pattern 71........
 session protocol sipv2
 session target ipv4:10.1.1.1
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 72000 voip
 translation-profile incoming SITEA
 session protocol sipv2
 incoming called-number 72........
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 200 voip
 destination-pattern 2..
 session protocol sipv2
 session target ipv4:10.1.1.1
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 500 voip
 destination-pattern 5..
 session protocol sipv2
 session target ipv4:10.1.1.1
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!

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