Dialogic® Media Gateway Installation and

Configuration Integration Note

1. Scope
This document is intended to detail a typical configuration of a Dialogic® 2000 Media Gateway
Series (DMG2000) when used with a Skype SIP trunk. This document does not however cover
the programming between the gateway and the PBX. For this information, please refer to the
Integration Notes posted on the Dialogic Service Center - Technical Helpweb:
http://www.dialogic.com/support/helpweb/mg/default.aspx

2. Configuration Details
Listed below are the specific details of the gateway and Skype Connect used in the testing to
construct the following documentation.

2.1 Dialogic® Media Gateway

Gateway Model Dialogic® 2000 Media Gateway Series

(DMG2030DTIQ)
Software Version Version 6.0 SU6 or later
Protocol T1/E1
PBX/Integration Any

2.2 Skype Connect SIP Trunk

Vendor Skype
Information Link http://www.skype.com/intl/en-us/business/sip/get-it-now

2.3 System Diagram

The diagram below details the setup used in the testing and creation of this technical document.

PBX Stations WAN/Public IP

SIP Trunk(s)
Skype

T1/E1

Dialogic® 2000 Media

PBX Gateway Series
(DMG2030DTIQ)

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3. Prerequisites

3.1 PBX Prerequisites

This document assumes that PBX programming or Direct PSTN connectivity has been
established by using other published PBX-specific configuration guides.

3.2 Gateway Prerequisites

The gateway used in this document is a Dialogic® 2000 Media Gateway Series (DMG2030DTIQ)
model running software Version 6.0 SU6, but this configuration can apply to all Dialogic® 2000
Media Gateways running Version 6.0 SU6 or later. External Client IP Address, Client Subnet
Mask, Default Network Gateway Address, and DNS Server Address will be needed for this setup.

4. Skype Connect Setup Notes

In order to successfully connect the gateway to a Skype SIP Connect trunk, the following steps
need to be taken:

1. Sign up for Skype Manager.

2. Buy Skype credit.
3. Set up a SIP profile (profile details are needed for successful gateway setup).
4. Buy online number(s) and assign it to the SIP Profile (for inbound testing).
5. Buy a local and long distance subscription and assign it to the SIP Profile (for outbound
testing).
6. Allocate credit to the SIP Profile.

More details on signing up for Skype Connect are available here:

http://www.skype.com/intl/en-us/business/sip/get-it-now

Once you have successfully signed up for Skype Connect and created a SIP Profile, go to the
Authentication Details tab under your profile settings and ensure that the method of
authentication is set to Registration. Under Registration, the following parameters will be listed:

• SIP User
• Password
• Skype Connect Address
• UDP Port 5060

Important: Make a note of these parameters as they will be needed when configuring the
gateway.

5. Gateway Setup Notes

5.1 Connection
1. Connect gateway to Local Network using LAN 1.
2. Configure computer connecting to gateway on the 10.12.13.x subnet (e.g. 10.12.13.75) and subnet
mask of 255.255.255.0.
3. Login to the gateway via web browser at: http://10.12.13.74 and start configuring gateway
for Skype Connect.

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5.2 Parameter Configuration

5.2.1 Configuration -> IP

- Once logged in to the gateway, the Client IP Address, Client Subnet Mask, and
Default Network Gateway Address need to be changed to the external ones.
Set to External IP Address

Set to External Subnet Mask

Set to External Network Gateway

- Submit changes and restart the gateway.

- Using a computer with internet access, point your web browser to the gateway’s
external IP address in order to complete the rest of the configuration.

5.2.2 Configuration ->TDM

In this document, a T1 ISDN interface was used. A screenshot of its configuration is
provided for reference.

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5.2.3 VoIP -> General
- User-Agent -> Host and Domain Name = External IP Address.
- Server -> DNS Server address = Primary DNS Server IP Address.
- Server -> DNS Server Address (Secondary) = Secondary DNS Server IP Address.
- TCP/UDP -> UDP/TCP Transports Enabled = Yes.
- TCP/UDP -> TCP/UDP server Port = 5060.
- Submit changes.

Set to External IP Address

Set to Primary DNS Server IP Address

Set to Secondary DNS Server IP Address

Set to Yes

Set to 5060

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5.2.4 VoIP > Network Groups
- Add or modify an existing Network Group.
- Network Group-> Network Group Label = SKYPE (Network Group SKYPE was
created earlier in Section 5.2.4).
- Transport ->Transport Protocol = UDP.
- Registration-> Registration Server Address = sip.skype.com.
- Registration-> Registration Server Port = 5060.
- Registered User = Your Skype SIP User (refer to Section 4 for details).
- Gateway Name = sip.skype.com.
- Submit changes.

Set to SKYPE

Set to UDP

Set to sip.skype.com

Set to 5060

Set to SKYPE SIP User

Set to sip.skype.com

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5.2.5 VoIP -> Authentication
- VoIP Authentication -> Outbound VoIP [Client] = Click radio button.
- Outbound VoIP Configuration -> Outbound Authorization Enabled = Yes.
Note: Two Realm entries need to be created next.
- Realms -> Realm = sip.skype.com.
- Realms -> User Name = Skype SIP User (refer to Section 4 for details).
- Realms -> Password = Skype SIP Password (refer to Section 4 for details).
- Realms -> Realm = Gateway External IP Address.
- Realms -> User Name = Skype SIP User (refer to Section 4 for details).
- Realms -> Password = Skype SIP Password (refer to Section 4 for details).
- Submit changes and reboot.

Set to Yes

Set to sip.skype.com

Set to Skype SIP User

Set to Skype Password

Set to External IP Address

Note: After the reboot, a successful registration and authentication with the Skype
Connect Service should take place. A gateway trace should be examined for a successful
registration confirmation.

5.2.6 Routing Table

Next, you will set up the routing rules needed to direct incoming DID Skype numbers into
the PBX and outgoing calls from the PBX utilizing the Skype SIP trunks.
Some of the parameters in the routing rules included in this section are specific to the test
scenarios described below and might require modification depending on your specific
PBX dial plan.

Numbers used for this testing:

Skype to PBX

Skype DID Online Number Target Destination

(Purchased from Skype) PBX Extension
7165687714 Extension 7714

PBX to Skype

PBX Trunk Access Code Destination #

8 1xxxxxxxxxx

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 Test Scenarios:

Outbound
User 1

716-555-1212
Extension
Skype
7714
T1 Sip Connect Trunk
DMG2000

PBX
User Dials 817165551212 | PBX Directs call to DMG | DMG Strips the 8 and dials 17165551212 | Skype Dials 7165551212

Skype Online #
716-568-7714
Inbound User 1

Extension
Skype 7714
Sip Connect Trunk T1
DMG2000

PBX

Someone calls 716-568-7714 | Skype Directs it to DMG | DMG Checks Routing table, Translates, and Sends it to the PBX | PBX sends to matching Ext

In order to accomplish the test scenarios described above, the following routing
configuration is needed:

- Router Configuration -> VoIP Host Group.

- Add a VoIP Host Group and name it SKYPE.
- In the Network Group drop down list, select SKYPE (Network Group SKYPE was
created earlier in Section 5.2.4).
- Assign sip.skype.com in the Host List.
- Submit changes.

Set to SKYPE

Set to sip.skype.com

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- Router Configuration -> TDM Trunk Groups.
- Ensure that TDM Channels are available.

- Router Configuration -> Inbound VoIP Rules.

- Depending on your PBX setup and routing needs, enter the appropriate manipulation
parameters. In this example, there is an Inbound VoIP Rule that accepts all Inbound
VoIP Traffic, extracts the right most 4 digits of the called party; rext(D,4), and
forwards them to the PBX.
- Submit changes.

Set to TDM

rext(D,4) extracts the right most

4 digits

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- Router Configuration -> Inbound TDM Rules.
- Depending on your PBX setup and routing needs, enter the appropriate manipulation
parameters. In this example, there is an Inbound TDM Rule that accepts all Inbound
TDM Traffic, extracts the right most 11 digits of the called party; rext(D,11) and
forwards them to Skype Connect. For example, if 817165551212 were dialed, the
gateway will only send 1716551212 to Skype Connect.
Important: Under Outbound Routes -> CPID Manipulation -> Calling Party Number,
the Skype SIP User (refer to Section 4 of this document) must be entered. Without
this entry, TDM to Skype calls will not work.
- Submit changes.

Set to Skype SIP User

rext(D,11) extracts the

right most 11 digits

This concludes the gateway configuration needed to interface with SIP Connect trunks.
You should now be able to execute the test scenarios included in this document.

Note: Outbound calls from the PBX through the Skype Connect SIP will display the
calling number set in your Skype Profile under “Caller ID” (Online Number, Landline
Number, or None). Please refer to Skype Connect documentation for further details on
Caller ID support.

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6. Troubleshooting

6.1 Important Debugging Tools

• Ethereal/Wireshark – Used to view and analyze the network captures provided by the
Dialogic® gateway diagnostic firmware.
• Adobe Audition – Used to review and analyze the audio extracted from the network
captures to troubleshoot any audio-related issues.

6.2 Important Gateway Trace Masks

These keys are helpful during all troubleshooting scenarios and should be considered keys to
activate by default for all troubleshooting cases.
• voip prot and voip code – this allows the collection of all SIP-related messages as
they are sent from and received by the gateway. This data is important in cases where
you feel that the gateway is not able to communicate properly with the messaging server.
• tel event and tel code – This allows the collection of all circuit-side activity of the
emulated station set such as display updates, key presses, light transitions and hook
state changes. This data is very important in the following scenarios:
o Call control problems (dropped calls, failing transfers, etc…)
o Integration problems (incorrect mailbox placement, missed auto-attendant
greetings etc…)
• teldrv prot – This allows the collection of all ISDN messages both transmitted and
received on the gateways front-end interface. This data is very important in the following
scenarios:
o Call control problems (dropped calls, failing transfers, etc…)
o Integration problems (incorrect mailbox placement, missed auto-attendant
greetings etc…)
• Routingtable (all keys) – This allows you to look inside the routing table engine
and see how matching rules and CPID manipulation rules work with respect to your call.
This data is very important in the following scenarios:

Note: Turning on all traces is not recommended. Doing this floods the debug stream with
significant amounts of information that can cause delays in determining the root cause of a
problem.

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05-2697-001
October 2010

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