ISDN VoIP VPN Link Info

[TimDimond]

I have an IP Office 406 (1.4(22)) with 3 x 30 POT modules (3.4(22)), 1 ETSI PRI and LAN connection at our central site. At a small remote office I have an IP Office 401 (1.3(34)) with 1 BRI (ETSI, BT Highway) and LAN connection.

At the moment the two are linked with a VoIP 'line' via ADSL connections. Both sites have voice networking enabled and so learn extension information from the other. All calls from the smaller site are routed via the VoIP line to cut down on call costs and 'anonymise' calls. This is working OK, but the line drops out quite frequently as a result of ADSL congestion.

I have just had Home Highway - ETSI ISDN - installed at the smaller site with a package that gives us unlimited free calls of up to an hour in length - as a registered charity keeping costs to a minimum is very important to us. Basically I need to set up the swithes so that they continue to work the same way as they do using the ADSL VoIP line, but using the ISDN line as the data bearer to improve call quality.

I have set up a service, line and IP route at the smaller office, and the matching RAS, user, line, incoming call route and IP route at the larger. The smaller office initiates an ISDN call to the larger office but it looks like this is rejected for some reason I don't understand.

Can anyone give me some additional hints and tips because I can't think of anything else to try? Also, is this the best way of setting up this link or is there a better alternative?

many thanks - Tim.

 

[routermad]

To help you decide what you actually require, try searching for a trail/full version of network monitor applications. this should help you determine what is happening on your network.

Does your ADSL router support QoS? If it does then great switch this on, other wise you may require additional hardware. I wont bore you with QoS definition, look it up in any computer/telecom dictionary.

With regards to your configuration, ask your maintainer to review it. If not I'm sure there are plenty of other supplier who would be happy to look after your needs. Are you based in the UK?

 

[MrIPO]

Hi have attached the IPO ISDN configuration guide for configuring a point to point link between IPO's via ISDN

Please note each call with comsume approx 26.4kbps. Thats using G.729a @ 20MS over a PPP session with no header compression. This will differ if you are going to use MLPPP to bond two channels together. if you are going to do this disable MPPC on the services on both the IPO's, with MLPPP the bandwidth will be 28kbps per call

You need to complete both the SITE A And SITE B configurations. Changes the settings to reflect your setup requirments eg: subnets, usernames and passwords

SITE A
1. In the System configuration form, change the IP address of LAN1 to 192.168.42.1.
2. Save and send the configuration to the IP Office.
3. Release and renew the IP configuration on your PC.
4. Create a new Normal Service as follows:
A) Name = ACME
Account Name = California
Password = Bells
Confirm Password = Bells
Telephone = phone number at Site B
B) Create a new IP route as follows:
IP address = 192.168.45.0
IP mask = 255.255.255.0
Destination = Georgia
C) Check the following default Incoming call route is
configured correctly:
Incoming number = blank
Incoming bearer = Any Data
Destination = DialIn
D) Check the following default RAS service is
configured correctly:
Name = DialIn
E) Create a new User as follows:
Name = ACME
Password = Benskins
Confirm Password = Benskins
Dial In On = Ticked
5. This User will now be automatically linked to the Service called ACME, as they both have the same name (the User Password is now the Incoming

SITE B
1. Check the following default Incoming call route is configured correctly:
Incoming number = blank
Incoming bearer = Any Data
Destination = DialIn
2. Check the following default RAS service is configured correctly:
Name = DialIn
3. Create a new User as follows:
Name = California
Password = Bells
Confirm Password = Bells
Dial In On = Ticked
4. Create a new IP route as follows:
IP address = 192.168.42.0
IP mask = 255.255.255.0
Destination = California
5. Create a new Normal Service as follows:
Name = California
Account Name = ACME
Password = Benskins Confirm Password = Benskins
Telephone = phone number at Site A
6. This Service will now be automatically linked to the User called California as they both have the same name (The User password is now the Incoming Password).
7. Save and send the configuration to the IP Office.
Both Sites

1. When both sites are ready, test the connection by pinging one another (If you are at Site A, ping B. If you are at Site B, ping A.).

Hope this helps

 

[TimDimond]

Thanks a lot MrIPO,

I'm just following your instructions now. There is one point though, should the IP route destination at Site A actually be ACME (the same name as the service and user)? As the destination is on a drop-down list I can't enter a value that has not already been defined on the system.

Take care,
Tim.

 

[TimDimond]

Another couple of questions for MrIPO (sorry, forgot these).

I presume that your instructions create an IP data link between the two sites using ISDN? This being the case I presume I need to create a new VoIP line at each end in order to route voice calls over this link? Should I use the following settings for this?

1. Gateway = the IP address of the remote site
2. Compression Mode = G.729a 8k CS-ACELP
3. H450 Support = H450
4. Tick Silence Suppression
5. Tick in Enable Faststart
6. Tick in Fax Transport Support
7. Tick in Out of Band DTMF
8. Tick in Voice Networking

Once this is done do I need to add any short codes to make sure calls are routed via this link? Site A is the primary site and has PRI installed - so I presume it will not need any. Site B has the BRI Highway connection and should be able to route calls to extensions at Site A via the VoIP link - releasing their extension number as CLI. All other other calls should be made via the ISDN line directly keeping CLI anonymous (witheld). What short codes do I need at Site B in order to achieve this.

Thanks a lot,
Take care,
Tim.

 

[MrIPO]

Sorry for the delay

I'm just following your instructions now. There is one point though, should the IP route destination at Site A actually be ACME (the same name as the service and user)? As the destination is on a drop-down list I can't enter a value that has not already been defined on the system.

The IP Route points to the service that you have created on each site, this would allow the destation network to be found via the connection that the service has outlined in its configuration

1. Gateway = the IP address of the remote site
Correct
2. Compression Mode = G.729a 8k CS-ACELP
Correct if you want approx 26.4kbps to be comsumed
3. H450 Support = H450
Yes for features
4. Tick Silence Suppression
NO
5. Tick in Enable Faststart
NO
6. Tick in Fax Transport Support
Only if you want to be able to fax over the VoIP Line
7. Tick in Out of Band DTMF
YES
8. Tick in Voice Networking
YES

Once this is done do I need to add any short codes to make sure calls are routed via this link? Site A is the primary site and has PRI installed - so I presume it will not need any. Site B has the BRI Highway connection and should be able to route calls to extensions at Site A via the VoIP link - releasing their extension number as CLI. All other other calls should be made via the ISDN line directly keeping CLI anonymous (witheld). What short codes do I need at Site B in order to achieve this.

You would need to make sure that the VoIP lines do not have the same outgoing group ID as the normal ISDN connections. You would for example give both lines ID 10 and create short codes going out though Line Group 10 and Not line group 0 which is the standard line group for outgoing ISDN calls

Let me know what number ranges you want to route over this link and i can help with the short codes.

Hope this helps

 

[TimDimond]

Hello again MrIPO,

The number ranges I need routing via the new VoIP Link are as follows:

At site A - Just the numbers 790-797 inclusive.

At Site B - 100-789 inclusive (with the exception of 500), 800-998 inclusive, any number starting with 118 and *604 (a short code to check a single voice mail box at Site A) - all these numbers need to send the originating extension number using CLI. All other Site B calls need to be routed through the ISDN line but witholding the CLI of the caller.

Many thanks for your on-going assistance.

Take care,
Tim.