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VoIP over an IPSEC Tunnel configured but not working...

femi.agboade
Level 1
Level 1

Hello,

I have some challenges with a VPN setup for voice traffic I recently configured for a client.

I have at the HO (Head Office) the following:

- 1800 router

- Avaya phones and Gateway

- 1MB radio internet access

At the BO (Branch Office), I have:

- 871 Router

- Avaya phones

- 256k internet bandwidth

The only reason we setup the VPN was setup was to enable the phones at the BO to connect to the gateway at the HO while also being able to make and recieve calls at the BO.

The phones at the BO successfully register to the HO, but are unable to recieve calls and dial out. Everytime I try to make a call, the phone displays a "connecting..." message.

Please see the attached config files.

Could it be the bandwidth of the VPN tunnel because when i checked sh int tunnel, i got the message below:

Tunnel0 is up, line protocol is up

  Hardware is Tunnel

  Description: Tunnel to HO

  Internet address is 192.168.200.2/30

  MTU 1514 bytes, BW 9 Kbit/sec, DLY 500000 usec,

     reliability 255/255, txload 1/255, rxload 1/255

  Encapsulation TUNNEL, loopback not set

  Keepalive not set

  Tunnel source 2.2.2.2, destination 1.1.1.1

  Tunnel protocol/transport IP/IP

  Tunnel TTL 255

  Fast tunneling enabled

  Tunnel transmit bandwidth 8000 (kbps)

  Tunnel receive bandwidth 8000 (kbps)

  Last input 00:00:13, output 00:00:13, output hang never

  Last clearing of "show interface" counters never

  Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 1419

  Queueing strategy: fifo

  Output queue: 0/0 (size/max)

  5 minute input rate 0 bits/sec, 0 packets/sec

  5 minute output rate 0 bits/sec, 0 packets/sec

     2376 packets input, 293066 bytes, 0 no buffer

     Received 0 broadcasts, 0 runts, 0 giants, 0 throttles

     0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort

     4453 packets output, 634702 bytes, 0 underruns

     0 output errors, 0 collisions, 0 interface resets

     0 output buffer failures, 0 output buffers swapped out

Any suggestions please will be very welcomed. This matter is becoming rather embarassing now

Regards,

Femi

3 Replies 3

Brian Meade
Level 7
Level 7

Try setting up a site-to-site VPN instead.  You also did not add any deny statements to your NAT access list so all traffic is still getting NAT-ed.

Hi Brian,

What I have in the config is a site-to-site VPN. Kindly go through my configs again.

However, your comments on the NAT access list sounds like somewhere to look into again. I will make the necessary changes and revert.

Thanks.

Femi

Hi again Brian,

I made a few changes but result is still same. Phones at the branch office still get registered, but they dont get a dial tone when you pick up the handset, neither do they ring out when they are dialed. Please see attached new configs. Any thoughts?

Regards,

Femi

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