How to setup Freeswitch with DIDforSale

Sep 22, 2014 | HowTo, SIP TRUNKING, VoIP Getting Started

How to setup Freeswitch with DIDforSale

You need to allow our IPs in your freeswitch server to receive calls from the Phone number. For that, go to your freeswitch configuration directory and edit the file acl.conf.xml inside the autoload directory of freeswitch configuration directory. Find the domains section in the configuration file and add our IPs as shown below

<node type=”allow” cidr=”209.216.15.70/28″/>

<node type=”allow” cidr=”209.216.2.211/26″/>

Now you need to add a dialplan entry in your server to catch the incoming calls to the Phone number. For that go to your freeswitch configuration directory, then move to dialplan/public directory and add an xml file there with any name with the below contents.

<include>

<extension name=”incoming_did”>

<condition field=”destination_number” expression=”^(13043935064)$”>

<action application=”transfer” data=”1000 XML default”/>

</condition>

</extension>

</include>

NOTE: Replace 13043935064 with your Phone number

This dialplan will route all incoming calls coming to the phone number to extension 1000

You will need to reload your acl and xml after doing the configuration. For doing that, go to freeswitch cli by using command “fs_cli” and run the commands reloadacl and reloadxml.

Please test it and let us know if you where not able to get this working. Feel free to contact us at contact-support@didforsale.com

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Step by Step How to setup SIP trunks in Asterisk?

Sep 18, 2014 | SIP TRUNKING

*Try Risk Free for first month and know the DIDforSale SIP Trunking difference*

asterisk sip trunk provider

 Some of our key SIP Trunking features:-

✓ Scale as per your need
✓ Flexible Pricing options to choose
✓ Reliable service & support
✓ Easy account management
✓ Multiple compatible platforms

✓ Largest Phone Number inventory
✓ SMS/MMS enabled Phone Numbers
✓ Number Portability
✓ Easy Integration
✓ 30-day money back guarantee

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Step by Step How to setup SIP trunks in Asterisk?

To setup the SIP trunks in your Asterisk machine is quite an easy job if you are using DIDforSale as your SIP provider. We will explain this process step by step:

A) Creating the SIP Trunks for Inbound service:

Step 1:

Login to your Asterisk PBX admin interface, go to Connectivity tab and click on Trunks and select the option of Add SIP Trunk and then give a name for the trunk as didforsale_1 and add the trunk Parameter as shown below:

host=209.216.2.211

type=peer

context=from-trunk

disallow=all

allow=ulaw

nat=yes

canreinvite=yes

insecure=very

dtmfmode=rfc2833

qualify=yes

Save and Apply the changes.

Step2 :

Create a second trunk for inbound with trunk name as didforsale_2 and add the trunk Parameters as shown in below:

type=peer

nat=yes

insecure=very

host=209.216.15.70

dtmfmode=rfc2833

disallow=all

context=from-trunk

canreinvite=yes

allow=ulaw

Save and Apply the changes.

Step 3:

Create Inbound routes.

For routing the inbound calls coming on your DID number, click on Inbound routes of your PBX interface and configure the DID with prefix 1. Toll free numbers needs to be configured without 1.

Say your DID is 9494568787  then you will configure the DID with 19494568787 in the inbound routes. It means that DID should be added with numbers only. No signs like ‘-‘are allowed in between the digits of DID number. Save and Apply the changes.

2) Creating the Trunks for Outbound Service:

Step 1:

Go to Connectivity and Click on Trunks and select the option Add SIP Trunk. and then give a name for the trunk as didforsale_out and add the trunk Parameter shown below:

type=peer

nat=yes

insecure==very

host=209.216.2.212

dtmfmode=rfc2833

disallow=all

context=from-trunk

canreinvite=yes

allow=ulaw

Step 2:

Create a second trunk in the same way with trunk name as didforsale_out2 with the below trunk parameters

type=peer

nat=yes

insecure=very

host=209.216.15.71

dtmfmode=rfc2833

disallow=all

context=from-trunk

canreinvite=yes

allow=ulaw

Step 3:

Create Outbound Routes

Click on Outbound Routes option of your PBX interface. In Dialed Number manipulation rules, put 1X.  in Match pattern column for US and Canada calls. Select the both
Outgoing trunks and Save and Apply the settings.

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when you can get for free.

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Visit SIP Trunking Pricing to see which plan suits your business!

With so many options to pick from it can often be hard to decide what’s best.
Our plans have been packaged together to give you optimum output.
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Our SIP Trunks are Compatible with wide range of PBX & Platforms.

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Unlimited SIP Trunking

Sep 12, 2014 | News, SIP TRUNKING

We are announcing that now you can purchase two way unlimited SIP Trunks from DIDForSale.

DIDForSale leader in SIP trunking services, offers the largest Coverage in US, UK and Canada offering more than 11,000 rate centers.

Why to consider Unlimited SIP Trunking

  • With two way SIP Trunking you still get unlimited origination from all over the world, you can also make unlimited outbound calls to Canada and US lower 48 states.
  • TwoWay SIPTrunk is perfect solution small to large size Businesses with in house PBX and also for personnel use, where you don’t have to worry about per minute charges. SIPtrunks can be shared between multiple employees and DIDs. (Fair use policy applies).
  • Two-way SIPtrunks can not be used for Call Centers, Conferencing, Calling Card and automated dialer applications. This product must be used strictly for conversational traffic.
  • To add two way SIP Trunking, please login into your account with DIDforSale and Select “Two Way SIP Trunks”.
  • You can still get local numbers from anywhere in US, UK and Canada on the same trunk.
  • We allow one to many relations. That means you can get 10 SIP Trunks and 100 Virtual phone numbers from anywhere in US, UK and Canada.

More Information on SIP Trunking

What is SIP Trunking

How SIP Trunking Works

SIP Trunking Pricing

Which SIP Trunking should I Choose

If you need help and want to talk to us, please send us an email or contact us.

Business Development Team

Email: contact-sales@didforsale.com

SIP Trunking for dummies

Aug 15, 2014 | SIP TRUNKING

Sip Trunking for dummies

Ok, we have been talking about SIP trunking for last few years. Keep bragging about it, SIP Trunking, is cool it saves business up to 80% on monthly bills. While talking to our sales team we learned that most of people call for VoIP phone lines, don’t understand how does SIP trunking actually works. Most of them recently purchased VoIP PBX system for almost around 10K. Looking at everything it seems like only tech savvy people are really taking the benefit of SIP Trunking. How about millions of non tech businesses owners. Unfortunately they are still paying lot more than they should. Here is how and why.

Let’s make some assumption and take an example of a small business

  • 100 employees in a consulting company
  • assuming 40% sales and support, primary telephone users,
  • 60% workers, researchers or developers, low to normal phone usage.
  • Business has SIP based telephone system, proprietary or open source like, Asterisk and Freeswitch.
  • Business has VoIP phones connected to PBX.

So if this business goes to local telephone company like cox or Comcast or ATT. They will be paying around $50 per line, so total telephone bill will be around $5000. That should give Unlimited local and long distance calling for each employee. You can get phone numbers only from local area.

With SIP trunking for the example above, you can evaluate how many people will be on phone at the same time. Let’s say 50%. Here the saving start

  • Why to pay for 100 telephone lines when your business is only going to use 50 SIP trunks at peak time.
  • Why not to choose a company which can provide you local numbers anywhere in the country. Getting local phone numbers in different areas is becoming very important, especially for sales people. When calling for new leads, there are more chances to convert the leads in sales if they see the local calling number.
  • Why to pay $50 per line when you can get the same service for half the price.

DIDforSale provides SIP Trunking for business all over US, UK and Canada. You can get a pool of SIP trunks and get phone numbers from anywhere. Learn more about SIP Trunking and how you can save on SIP Trunking with DIDForSale.

How to Enable CNAM on DID

Jul 7, 2014 | HowTo, SIP TRUNKING

DIDForSale leader in Origination and Termination SIP trunking services, offers the largest Coverage in US, UK and Canada, now offers E911 service for the DIDs purchased through us.

Here are easy steps to enable CNAM to your DIDs

  • Login in your webportal at www.didforsale.com
  • Click on “Add CNAM Service”
  • You will see the list all your VoIP DIDs, assigned to your SIP trunk. Select the VoIP DID you want to enable CNAM on.
  • CNAM service must be enabled for each DID.
  • Click on Confirm.

We have two different plans for CNAM service, metered and Flat Rate. For All the DID enabled for CNAM, we add the caller ID name to each call. If the phone number does not have registered caller ID name, we get City and State as Caller ID Name. In case of international calls we send Country Name as CNAM field.

In Metered CNAM Service you pay per query. For every call you receive you will pay flat fee. In Flat Rate, you pay only $1 per DID and get unlimited Queries (Fair Use policy apply).

Feel free to contact us for any questions or concerns.

DIDFOrSale.com
WholeSale SIP Trunk Provider

* Flatrate CNAM service is available only for normal business use and cant be used for High Volume Inbound calls for Call Center and Calling Card.