SIP Trunking using the EdgeMarc Network Services...

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Property of Cox Communications, Inc. Version 0.3

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July 9th, 2013

SIP Trunking using the EdgeMarc Network

Services Gateway and the Trixbox IP-PBX

12.6

© 2011, Cox Communications, Inc. All rights reserved.

This documentation is the confidential and proprietary intellectual property of Cox

Communications, Inc. Any unauthorized use, reproduction, preparation of derivative

works, performance, or display of this document, or software represented by this

document is strictly prohibited.

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Table of Contents

1 Overview .............................................................................................................. 2

2 Prerequisites ......................................................................................................... 3

3 Network Topology .................................................................................................. 3

4 Description of Basic Operation and Call Flows ............................................................ 4

5 PBX Configuration .................................................................................................. 4

1 Overview

The purpose of this configuration guide is to describe the steps needed to configure the Asterisk PBX for proper operation in a SIP Trunking application with the e-SBC EdgeMarc. Please note that this guide documents the basic configuration needed in the Asterisk PBX and that the requirements of specific SIP Trunking environments may require modifications to the configuration steps provided in this document.

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2 Prerequisites

SIP Trunking information provided by the VoIP service provider:

SIP proxy server IP address or DNS name.

Trunking Direct Inward Dial (DID) phone numbers o Calls to the Trunking DID(s) are forwarded from the service provider to the

wide area network (WAN) IP address of the EdgeMarc. There may be a single “Pilot” phone number used for all inbound calls and/or multiple DIDs depending on the service provider settings.

SIP authentication credentials (optional) o Some SIP Trunking service providers require a unique username and

password to be supplied for IP PBX registrations and/or SIP signaling using P-Asserted-Identity (RFC 3325). This configuration guide provides the configuration steps for both PBX registration and static or non-registration modes of PBX operation.

3 Network Topology

Figure 1 Test Set up

The PBX in the above network topology represents the PBX that is connected via its LAN port to

the LAN port of the EdgeMarc Network Services gateway.

Table 1 – PBX Information

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Manufacturer: Fonality

Model: Trixbox Standard Edition

Software Version: 12.6

Does the PBX send SIP

Registration messages (Yes/No)? Yes

Vendor Contact:

Table 2 – E-SBC Information

Manufacturer: Edgewater Network, Inc.

Model: 4552

Software Version: 11.6.14

4 Description of Basic Operation and Call Flows

Basic Call Flow:

All phones connect to the PBX. The PBX will interface with the service provider using SIP trunks.

Internal calls: Calls between phones on the LAN

LAN phone PBX LAN phone

Outbound calls: Call is initiated by a LAN phone to a WAN phone.

LAN phone PBX <SIP trunk> EdgeMarc SIP trunk service provider WAN phone

Inbound call: Call is initiated by a WAN phone to a LAN phone.

WAN phone SIP trunk service provider EdgeMarc <SIP trunk> PBX LAN phone

5 PBX Configuration

The steps below describe the minimum configuration required to enable the PBX to use SIP trunk

for inbound and outbound calling. Please refer to the PBX product documentation for more

information on SIP trunking or other advanced PBX features.

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The configuration described here assumes that the PBX is already configured and operational with

station side phones using assigned extensions or DIDs.

The PBX Ethernet port should be connected to the EdgeMarcs LAN port, and the LAN port to the

phones (IP address=192.168.2.1/24) on the same LAN subnet.

1. After the PBX software/OS has been installed onto the hardware system, it will need to be

configured with an IP address to reach the Internet. This is done via the DHCP server. Let

the PBX grab the IP from the EdgeMarcs DHCP IP pool, enter in the Username and

Password provided by the PBX provider and it will connect to the portal.

2. From the portal, login to access the PBX configuration menus.

Figure 2 Login

The Configuration Menu should appear:

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Figure 3 Configuration Menu

3. Reconfigure the Network Settings, hover over the Options Tab, then click the network link.

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Figure 4 Network Settings

Fill in each field appropriately.

IP Address Mode: Set to “Static”

Revert If Error: Set to “yes”

IP Address: 192.168.2.11

Subnet Mask: 255.255.255.0

Gateway: 192.168.2.1

DNS Mode: Set to “Static”

Primary DNS: Set to ISP provided DNS IP

Secondary DNS: Set to ISP provided DNS IP

DNS Forwarder: blank

Public IP Auto-detection: Set to “Off”

Public IP Address: blank

4. To configure the SIP Server and Registration Details, hover over the Options Tab, then

click the voip link.

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Figure 5 SIP Server & registration

5. Fill in the fields accordingly.

a) Route Name: Give the VoIP account a name to identify it by.

b) Username: Fill in the Pilot DID provided by the ISP

c) Password: Fill in the Pilot DID’s registering Password

d) Provider: Set to “Other” for the first drop down box, set to “SIP” for the second drop

down box.

e) Register: Set to “yes”

f) Server: Set to the LAN side IP address of the EdgeMarc

g) From User: Set to the Pilot DID

h) Direction: Set to “both – (friend)”

i) From Domain: Set to the IP address of the Trixbox

j) TOS: Set to “none”

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k) Authentication: Set to “plaintext”

l) Allowed IP’s, Disallowed IP’s, and Outbound Proxy leave blank

m) Codec 1 set to “ulaw”, Codec 2 set to “alaw” and Codec 3-4 set to “none”

n) Transfer: Set to “no”

o) SIP Re-Invite: Set to “yes”

p) NAT: Set to “no”

q) Insecure: Set to “very”

r) DTMF Mode: Set to “RFC2833”

s) RFC2833 Fix: Set to “no”

t) Qualify: Set to “none”

u) Send RPID: Set to “no”

v) Register String: This will auto populate.

Click the “Update VoIP Account” button to apply the settings.

6. To configure Extensions, Phones, and DID’s, select the Users / Extensions tab, click the

phone numbers link.

Figure 6 Update VoIP account

a) Number: Fill in the DID to be added, the first DID to be add must be the Pilot DID.

b) Type: Set to “VoIP”

c) Verify: Uncheck

d) Primary: Set to “yes” only for the Pilot DID (Or the DID to go to the Auto Attendant),

set to “no” for all others.

e) Select add user to configure the extensions for each phone. Select the “I want to add a

new user and a new extension” radio button.

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Figure 7 Add user

f) Configure the Extension accordingly.

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Figure 8 Add user

g) Fill in the Personal Information to match the Users information.

h) Under Voicemail Settings select the radio button “yes” and put in a password.

i) Under Routing and Appearance Settings it is possible to configure the Extensions Call

Forwarding. Under the “Inbound Phone No.” field select the DID to associate with the

Extension, or set to “none” if this is not needed. Under the “Outbound Caller-ID” field

it is possible to determine what Caller-ID should go out with the call. Select either the

Pilot DID (labeled as Global Default), Blocked (Anonymous), or a DID.

j) Under the “Primary Extension” set the desired Extension Number, set the DTMF Mode

to “RFC2833” and associate the Extension with a physical phone under the “Phones /

Devices” drop down menu.

k) Click the “Add User and Extension” button to submit the settings.

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7. To Configure the Auto Attendant, hover over the AutoAnswer Tab and click “edit call

menu” and modify as desired.

Figure 9 Edit call menu

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8. To change the Dial Plan, hover over the Options tab, click “dial plan”

Figure 10 Dial plan

a) Either click on an existing dial string or add a new one.

b) Under “Route” set the “1st:” drop down box to the correct provider.

c) Click “Update Dial Plan”

Figure 11 Update dial plan

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d) Repeat for each dial string.

The asterisk console can show the sip signaling that occurs between the phones and the SIP trunks, to enable this sip logging: *CLI> sip set debug on

If changes were made to the conf files, reload the plans: *CLI> reload

For advanced configurations and support please contact the Edgewater Technical Assistance Center [email protected] or call 408.351.7255.

mailto:[email protected]

tel:+14083517255

SIP Trunking using the EdgeMarc Network Services...

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