2801 - As VoIP Gateway

[norman_santana]

Hello Friends

I need to setup a Cisco Routers 2801 as VoIP Gateway

The scenario should be like this

|SIP Phone/trunk| -----> (ROUTER 2801) --------> | TDM e1 PRI ISDN | ----> PSTN

Any ideas that how i can configure this?

Regards

Norman Santana

[hiepnguyenho]

Hi Normal,

For your goal, you should follow these things:

1.Hardware:

- Router 2801 with voice supported IOS

- PVDM2-32 (minimum)

2. Configuration steps (these are the basic things to make a voice gateway works with CUCM or SIP PBX)

- Enable voice gateway function

- Configure E1 card and some ISDN parameters

- Configure sip trunk to SIP PBX. You said you want to connect SIP phone to Voice gateway, it is not possible. Voice gateway is just voice gateway, not the call agent for end point. If you want to register SIP phone to voice gateway, it has to be a CUCM Express.

- Configure H323 trunk from CUCM to your voice gateway.

- Configure dial plan (outbound, inbound call for PSTN, to extension number in CUCM system or SIP PBX system.. etc)

- Create some translation rule as in your dial plan

You should read about these things. If you need some example or guide, I will post.

Hiep Nguyen.

[norman_santana]

Hello Hiep

1.Hardware:

- Router 2801  with voice supported IOS - I  use c2801-advipservicesk9-mz.151-1.T.bin"

- PVDM2-32 (minimum) - Comply 2 DSPs, 32 Voice resources

The Follow I need the config sample

2.  Configuration steps (these are the basic things to make a voice gateway  works with CUCM or SIP PBX)

- Enable voice gateway  function

- Configure E1 card and some ISDN parameters

-  Configure sip trunk to SIP PBX. You said you want to connect SIP phone  to Voice gateway, it is not possible. Voice gateway is just voice  gateway, not the call agent for end point. If you want to register SIP  phone to voice gateway, it has to be a CUCM Express.

- Configure  H323 trunk from CUCM to your voice gateway.

- Configure dial plan  (outbound, inbound call for PSTN, to extension number in CUCM system or  SIP PBX system.. etc)

- Create some translation rule as in your  dial plan

Regards

Norman Santana

[hiepnguyenho]

Hi Norman,

I'm sorry for this lately reply, it's a hard working day.

I post some paragraphs here and attach the full configuration:

1. Configure ISDN line:

!

network-clock-participate wic 1

!

isdn switch-type primary-net5

!

controller E1 0/1/0
framing NO-CRC4
pri-group timeslots 1-31

!

interface Serial0/1/0:15
no ip address
no logging event link-status
isdn switch-type primary-net5
isdn incoming-voice voice
isdn T309-enable
isdn T306 3000
isdn T310 1000
isdn bchan-number-order ascending
no cdp enable

2. Enable voice gateway function

gateway
timer receive-rtp 1200

!

!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
cause-code legacy
h323
sip
! Specify the interface that makes a trunk with CUCM

interface FastEthernet0/0
description LAN Hutchison$ETH-WAN$
ip address 10.8.15.254 255.255.255.0 
h323-gateway voip interface
h323-gateway voip bind srcaddr 10.8.15.254

!

3. Enable sip connection

!

sip-ua

sip-server ipv4:10.30.25.252

! add username and password if your sip server needed

4. Configure dial-plan

!Outbound call

dial-peer voice 9 pots

description Outbound to PSTN
destination-pattern 9
progress_ind setup enable 3
progress_ind alert enable 8
port 0/1/0:15
forward-digits extra

!Inbound call to CUCM

dial-peer voice 2 voip
description Inbound to CCM
destination-pattern 2...
session target ipv4:10.8.15.25
dtmf-relay cisco-rtp
codec g711ulaw
no vad

!Call to SIP

dial-peer voice 9092 voip
translation-profile outgoing to092
destination-pattern 9092.+
session protocol sipv2
session target ipv4:10.8.14.32
dtmf-relay rtp-nte
codec g711ulaw

5. Auto Attendant

application
service inbound_ivr flash:welcome3.tcl

!Auto Attendant service

dial-peer voice 123 pots

service inbound_ivr

incoming called-number 35730...

There will be some different way to handle a call from PSTN to inside leg. You can use IOS for auto attendant and call handler or you can use Direct-inward-dial to directly call from PSTN to ext number.

That's all you need for basic Voice gateway operation. Post your request here for more advance function.