New topology for a remote site.

[Matthew Michaluk]

Please let me know if this is possible:

Call Flow

NEC PBX>>>>CME>>>>>>ASA5505>>>>>CAble Modem>>>>>>>>INternet>>>>>>>ASA5505>>>>>>>>MGCP Voice GAteway>>>>>NEC>>>>PSTN.

[Matthew Michaluk]

Nec>>>>PRI Connection to CME>>>>>>

MGCP>>>>>PRI Connection to NEC>>>>>PSTN

[t.s.davis]

I'm trying to get an idea of your overall topology. You list and MGCP Gateway. I assume there is a CUCM somewhere in the mix that the gateway is registered to?

Also, keep in mind, any voice traffic going across that Internet VPN link is going to be best effort with no QOS. 

[Matthew Michaluk]

yes the call manager is on the same vlan as mgcp gateway at the same location.

[t.s.davis]

You should be able to make this work. However, like I said, voice across an Internet VPN link is not typically recommended.

Just curious, why are you keepng the NEC around? My priority would be migrating off of the NEC and moving the PSTN connections over to the Voice gateway.

[Matthew Michaluk]

Cost of new IP Phones.   The main problem was a bad LEC connection with a Point to Point PRI connection from NEC to NEC.  Everytime it rains there is usually a problem with the connection.  WE will probably gradually migrate off the NEC to VOIP.

[Marwan ALshawi]

ok since the CME and MGCP are Cisco Routers

then i would recommend you to use some tunneling between your sites over the Internet

you can use a GRE tunnel from the CME to the MGCP and on the ASA use IPsec to encrypt this GRE tunnel so it will be ipsec over GRE over the Internet

GRE will be good to pass any required RTP media or multicasting traffic between your sites without having it inspected by the firewall ( good from VOIP point of view but not always good from security point of view ) and the IPsec will make sure you traffic over the Internet is encrypted

as stated by the above poster it is not always good to have your VOIP traffic with tunneling due to the added overhead

however if you have to this is an option for you

hope this help

if helpful Rate

[Matthew Michaluk]

Would you recommend just implementing a a Voice Gateway instead of a CME router?

[Matthew Michaluk]

I guess my biggest challenge is creating the dial peers from one PBX to another.

[t.s.davis]

Do you have any phones registered to the CME? If not, I would just convert it to an H323 or MGCP gateway. Personally, I would use H323. H323 will give you more intelligence at the gateway level and you won't be backhauling all of the control across the VPN to CallManager.

As for the dial-peers, from the NEC, you will need a VoIP dial-peer on the CME/VG to match the destination extensions on the other side. That VoIP dialpeer would point to the CallManager IP address on the other side of the VPN link.

[Matthew Michaluk]

What type of router would you recommend for 25-50 users?  This site will probably gradually move over to VOIP phones.

2901?

[t.s.davis]

I assume you are talking about for CME? If you want to support 50 CME phones, you will need to step up to a 2911. The Maximum phones for each platform is listed below. The data can also be found here:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/requirements/guide/cme90spc.html

2901 - 35

2911 - 50

2921 - 100

2951 - 150

3925 - 250

3925E - 400

3945 - 350

3945E - 450

[Matthew Michaluk]

I was just talking about connections from the VG and not CME or does it even matter if it will be a VG and not a CME?

[t.s.davis]

In that case a 2901 should be fine.

[Matthew Michaluk]

So as far as dial peers go I would need to point every thing to the tunnel and destination of the CUCM route out through the existing MGCP gateways.

1 is the dial out code.

dial-peer voice 30 voip

  description Used for 7-digit dialing areas

  destination-pattern 1[2-9]......

  progress_ind setup enable 3     

  voice-class codec 1

  voice-class h323 1

  session target ipv4:192.168.1.201

  dtmf-relay h245-alphanumeric

!

!

dial-peer voice 31 voip

  description Used for 10-digit dialing areas

  destination-pattern 1[2-9]..[2-9]......

  progress_ind setup enable 3     

  voice-class codec 1

  voice-class h323 1

  session target ipv4:192.168.1.201

  dtmf-relay h245-alphanumeric

!

dial-peer voice 32 voip

  description Used for 11-digit dialing areas

  destination-pattern 11[2-9]..[2-9]......

  progress_ind setup enable 3      

  voice-class codec 1

  voice-class h323 1

  session target ipv4:192.168.1.201

  dtmf-relay h245-alphanumeric

!

dial-peer voice 33 voip

  description Used for service numbers, such as 311, 411, 611 and 911.

  destination-pattern 1[3469]11

  progress_ind setup enable 3 

  voice-class codec 1

  voice-class h323 1

  session target ipv4:192.168.1.201

  dtmf-relay h245-alphanumeric

!

dial-peer voice 34 voip

  description Used International Dialing

  destination-pattern 1011T

  progress_ind setup enable 3

  voice-class codec 1

  voice-class h323 1

  session target ipv4:192.168.1.201

  dtmf-relay h245-alphanumeric