Question

Seting up a VOIP systme / Cisco UC500

Asked by: netcomp

We just got a Cisco UC500 system with 3 phones , but we are new to the VOIP. We have plenty of expertise in IT ( data)  
Currently we have 3 analog lines coming to our  office and we do not have a PBX phone system.
What want to figure out is that how the call routing does would work.  For instance, with email servers , you go to your ISP DNS and point the mx records to your public IP and thats how you get emails.
I am not sure how that would work with VOIP. Do we tell the phone company to forward our calls to our number to a certain IP.  And how would outgoing calls work, would they be just free long distance since we are using the internet.  I have read up on the user guides, but did not get a lot out. Thanks,

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Asked On
2008-09-05 at 10:35:46ID23707076
Topic

Voice Over IP

Participating Experts
2
Points
500
Comments
7

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Answers

 

by: bkepfordPosted on 2008-09-05 at 13:24:51ID: 22403413

If you are trying to replace your three analog lines with a VOIP telephone provider like Vonage or other carriers it all depends on how you want them to hand that to you. Ussually it is handed to you using a sip trunk which is setup as a dial-peer on your US500.

Here is a website that lists VOIP service provider or you can google it which is what I did to get this website.

http://www.voip-info.org/wiki/view/VOIP+Service+Providers+Business+North+America

The UC500 has 4 analog ports that it can except and you can setup your system to be a key system (Every phone has a line for each of the 3 numbers) that way all three lines ring on all of your phones and you can choose which of the three lines to dial out on. Or you setup the phones as extensions that you dial out on and each phone number that is forwards in would choose to route however you see fit.

With only three lines most people find it convient to setup as a key system to allow full visibility of the lines.

Lastly if this is your first time setting up a UC500 using the Cisco Configuration Assistant is the recommended way of installing it.

http://www.cisco.com/en/US/products/ps7287/index.html

 

 

by: netcompPosted on 2008-09-05 at 13:37:29ID: 22403534

I am not sure what a sip trunk is and how it works? Does it send all calls to our numbers to our public ip that the phone system is connected to?

If we were to keep our verizon anolog lines, than we are suing voip anymore are we?

What is the disadvantage of a key system vs the other?

 

by: lrmoorePosted on 2008-09-05 at 16:47:16ID: 22404691

The UC500 plugs into 1 or all 3 of those analog phone lines.
No SIP trunks, no SIP provider to deal with.
The phone company does not have to do anything. When that analog phone number rings in, the UC500 will direct the call to the IP Phone that you have designated. All outbound calls from all 3 IP Phones can share the 1 line, or load-share all 3 lines.

 

by: bkepfordPosted on 2008-09-08 at 11:55:24ID: 22420350

You would be using IP telephony on the Local LAN but you would not be using VOIP to save any tolls or anything like that .

The difference between a key system and a PBX is that every incoming line is mapped to a "key" on your phone it is the best way to set it up in a small office of less then 7 people. key systems you give up control for simplicity.

Once you start get getting larger you would actually look at doing a PBX which instead of mapping outside lines directly to the phones you would give each phone an extension. Some extension would be for internal calling only and others would be able to be directly dialed. A PBX basically gives you full control over a larger office space that you really don't need with a small office.  

Here is the sip trunk configuration using the CLI but I would suggest that you do it through the Cisco Configuration Assistant. Here is a link to configure the UC 500.

http://www.cisco.com/en/US/products/ps7293/prod_installation_guides_list.html

!--- SIP Trunk Configuration ---
 
dial-peer voice 1 voip
description **Incoming Call from SIP Trunk**
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number .%
dtmf-relay rtp-nte
no vad
!
!
!
dial-peer voice 2 voip
description **Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 9........
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
no vad
!
!
!
dial-peer voice 3 voip
description **Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 9[2-9]..[2-9]......
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
no vad
!
!
!
dial-peer voice 4 voip
description **Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 9[0-1][2-9]..[2-9]......
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
no vad
!
!
!
dial-peer voice 5 voip
description **911 Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 911
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
no vad
!
!
!
dial-peer voice 6 voip
description **Emergency Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 9911
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
no vad
!
!
!
dial-peer voice 7 voip
description **911/411 Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 9[2-9]11
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
no vad
!
!
!
dial-peer voice 8 voip
description **International Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 9011T
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
no vad
!
!
!
dial-peer voice 9 voip
description **Star Code to SIP Trunk**
destination-pattern *..
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
no vad
 
                                              
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by: netcompPosted on 2008-09-13 at 11:12:50ID: 22468573

Thank you bkepford,  
It would be very kind of you to also help with two follow up questions,
Two questions:
1. When customers calls, I would like the recording to give some extentions options ( or numbers) and the caller would chose where to connect. for example. Press 1 ( or the extention number) for sales, and..... . Would a " keysetup" do that?

2. I realy would like to know how we could do a VOIP even outside of the LAN . How does that work? and how does that save on toll free and long distance?




 

by: bkepfordPosted on 2008-09-15 at 08:44:27ID: 22479794

1) Hmm I think the PBX setup lends itself more to an automated attendant (AA) as the first point of contact although it is possible just to setup in Key mode for the lines to ring to an AA on a ring no answer but it gets kind of mirky best to do a PBX if this is your desired setup. All and all the setup is pretty much defaulted to that anyway.

2) If you have multiple offices you can route calls over the internet without any problems. You can get a sip provider as your primary long distance provider. And their is even ways to setup a Skype gateway that may let you leverage skype in your infrastructure.

Here is a link for the Skype gateway. I have set it up before and done some neat things with it but it doesn't scale well. I would say for an organization your size may turn into a great asset.

http://www.stonevoice.com/Products/IPSoftwareSkypeGateway/tabid/989/Default.aspx

 

 

 

by: netcompPosted on 2008-11-16 at 12:59:43ID: 22971959

I am still not clar on  some of my qeustion, but it has been a while so I will close this qeustion maybe start a new one.

20120131-EE-VQP-002

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