Parlay
From Pbxnsip Wiki
Parlay make a number of ISDN products including Multiplexors, Least Cost routers and VoIP Gateways. Their VoXip range of ISDN Gateways work well with PBXnSIP.
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Initial Configuration
The VoXip 104 ships with 2 BRI ports (4 channels). One of the ports is configured in TE mode while the other is in NT mode. To make both ports TE mode (to connect to the ISDN network) you need to open the back of the unit and switch some jumpers, consult the user manual for instructions on how to do this. Once both ports are configured in TE mode they can be connected to the ISDN wall socket providing 4 voice channels for your PBX.
Trunk Groups & Voice Carriers
Parlay use the terms Trunk Groups, Carriers and Routing Tables.
A Trunk Group is a resource of channels, for example BRI0+BRI1 or SIPProvider1 etc. A Trunk Group is where calls arrive and leave the gateway.
A Carrier is used with the routing tables. A carrier delivers the call to a trunk group. For example, in the routing table when the criteria is matched (e.g. Destination=999) then it will "call" that carrier, the carrier collects the call and delivers it to a Trunk Group.
For a more detailed explanation of these terms and some good training guides please see their support website http://www.voxip.dk/Help/
SIP Configuration
I will configure the Parlay to act as a 4-Channel ISDN gateway for the PBXnSIP server.
In this example 10.1.0.9 is the IP of the PBXnSIP box, while 10.1.0.7 is the IP of the ISDN gateway.
The first step is to configure the SIP Provider (PBXnSIP).
From the SIP menu you want to create/edit the first provider.
Click Edit next to the first provider which will take you to the configuration screen
Fill in the fields to match your PBXnSIP configuration.
On the main SIP screen there is no requirement to configure SIP Accounts as we will make and receive calls simply using the IP address.
Trunk Groups
We will have two trunk groups, the "ISDN Main" trunk groups which connects us to the PSTN network and the "SIP Trunk Group" which connects us to PBXnSIP.
On the Trunk Groups screen, you want to group BRI0 + BRI1 to be members of the "ISDN Main" trunk group and the SIP Provider will be the sole member of our SIP Trunk Group.
Now the Trunk Groups are configured, we need to configure the voice carriers which will work with the routing tables to pass calls PBXnSIP.
Voice Carriers
You should have two Voice Carriers, ISDN Carrier 1 which our ISDN Trunk group is linked to, and SIP Carrier 1 which are single SIP Provider is linked to.
Voice Routing
The Voice Routing section is where you tell the gateway how to deal with calls originating from a given trunk group. For instance, you tell the gateway what to do when it receives an INVITE for 999 from SIP Trunk Group, or an inbound call from a given network on the ISDN Trunk Group.
You can do routing using MSN/DDI schemes on both called party and calling party numbers. To keep things simple, we'll route all calls originating from the SIP Provider (PBXnSIP) out on the ISDN, and all inbound calls on the ISDN to PBXnSIP.
On the Voice Routing page, we will create two routes.
Net_to_SIP routes calls from the ISDN Trunk Group to the SIP Carrier (SIP Carrier 1 => PBXnSIP) SIP_to_Net routes calls from the SIP Trunk Group to the ISDN Carrier (ISDN Carrier 1)
Net_to_SIP
Net_to_SIP handles the calls received on the ISDN Trunk Group (BRI0,BRI1 etc) and routes the calls to the SIP Carrier, which in turn will send an INVITE to PBXnSIP.
? signifies a wildcard entry, so the routing line basically says, anything received from the ISDN Trunk group fthen send it to the SIP Carrier.
The same applies for routing from PBXnSIP to the ISDN
After you have built your trunk groups, carriers, providers and routing tables the changes will need activated, tables compiled and config saved.
DDI/MSN Behavior
If you have DDI/MSN Numbering on your BRI circuit then the Parlay will send an INVITE with the Called-Party-Number (the number that was called) to PBXnSIP. For instance, if your MSN number is 441624111111 then the Parlay would send an INVITE 441624111111@pbxnsip.local - to match this to a hunt group I simple adding 441624111111 as an alias, or an alias for a specific user account. Likewise within the Sip_To_Net voice routing you can add a line to say where the calling-party-number is 201 (extension 201) then change the CLI to 441624111112 - this will then present the correct CLI on the ISDN. This is just one way of doing it, you can shift some of these steps onto PBXnSIP itself, so it presents the correct CLI info (via RPID etc).
PBXnSIP Trunk Configuration
The PBXnSIP configuration is very simple. Inbound calls ring a hunt group (250) which in turn calls a number of accounts. Outbound calls simply go the IP address of the ISDN gateway. By default, if the Parlay matches the default wildcard route it will send an INVITE to 0000 at PBXnSIP.
Notes
I had some issues with call progress tones when dialing in from the ISDN. I fixed this by using the Tweak feature on the Parlay (Configuration-Voice-Advanced-SIP Tweaks -> Do not use inband Ringback Tone from SIP Server => Ticked). From then on in, I got call progress tones when I dialled in.
I highly recommend doing the various tutorials and reading the numerous FAQs & Guides on the Parlay website.
