How to call using VoIP

Making calls using VoiceAll is flexible and easy

From VoIP phones

We recommend using Linksys, Grandstream, Cisco, Polycom IP Phones or any SIP or H.323 compatible IP PBX. Configure your IP Phone / PBX with the VoIP settings sent in your email and you are ready to make calls.

Usage & Troubleshooting

An example configuration of Linksys ATA / IP Phone

Step1

You must first determine what IP address it received. To do this

IP Phone- On the phone dial pad, press the Setup button on your IP Phone . Using the Up/Down Scroll buttons navigate to “Network” and press the left Soft Key labelled “Select”. The IP address of the phone will be displayed in the “CurrentIP” field. Write the IP address down for future reference.

ATA - Connect an analogue telephone to the Phone 1 or Phone 2 port of the Linksys ATA.
- Press **** (quickly press the star key four times). Wait until you hear “Linksys configuration menu.”
- Check DHCP by keying in 100, if DHCP is enabled, proceed to next step. If DHCP is disabled, key in 101 and then 1 to enable DHCP.
- Check LAN IP Address by keying in 210. IVR announces the current IP address of the LAN port. Write the IP address down for future reference.

Step2

Go to any browser equipped computer on your network and enter the address: http://IP ADDRESS/ where IP ADDRESS is replaced by the address that was given to you in Step1.

Step3

Click on "Admin Login" and then on “Advanced”.

Step4

Click on SIP and find the RTP Parameters section. Change the RTP Packet Size value to 0.020:

Scroll Down

Step5

Click on Regional, find the Miscellaneous section and change Time Zone to the appropriate one:

Scroll Down

Step6

OPTIONAL:

If you experience low audio or your callee can't hear you properly, click on Phone and in the Audio Input Gain(dB) section, change fields values to 6:

Scroll Down

Step7

Click on Ext 1 and do the following:
- In the General section, make sure the Line Enable field is Yes.
- Under NAT Settings, the NAT Keep Alive Enable field should be Yes.
- Under Proxy and Registration, key in Proxy: sip.voiceall.net and set Register: Yes.
- Under Subscriber Information set the values same as you received in your activation email.Display Name: username, User ID: SIP ID, Password: SIP Password

Finally, click on Submit All Changes

How to configure other IP Phones?

Whatever IP Phone you use, below are the settings you would need to configure your IP Phone with. Below settings are specific to each account. Please check your activation email for relevant setup of your own account
Example:
SIP ID / User Name: 1022
Password: xxxxxx
Proxy server: sip.voiceall.net
Port: 5060

Some IP Phones ask for Authorization Username, please enter your Username / SIP ID and a Domain, which is sip.voiceall.net If you need further assistance, please raise a ticket at Help Desk detailing the issue with the make and model of the IP Phone you are using.

How to configure Asterisk, Trixbox, FreePBX, Nortel or other IP PBX?

Common Connection Diagram: Any IP/PSTN-capable PBX can be connected to Voiceall MOR Soft Switch.

The main principle here is to define Voiceall MOR as a Trunk in any of the PBX’s you are going to configure. We are going to consider Trixbox of an example configuration.

As an example, we will configure one phone dialling. The phone is connected to Trixbox, which is connected to Voiceall MOR:

Here, for simplicity, all the devices are in the same LAN:
Voiceall MOR server has IP: 10.10.1.10
Trixbox: 10.10.1.20
Phone: 10.10.1.30

Voiceall MOR and Trixbox are connected with the following settings:
username: mor_trixbox
password: mor_trixbox_psw

The phone to Trixbox is connected with:
username: 201
password: 201psw

Trixbox Configuration for Voiceall MOR

In Trixbox we need to define Voiceall MOR as a Trunk

Define a route in Trixbox to use this Trunk (this is just an example)

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