3 Simple Ways to Improve ASR on your VOIP traffic

ASR (Answer Seizure Ratio) is a great way to measure your underlying VoIP carrier’s network performance.  With that said, there are many other factors that affect your ASR.

First let’s discuss what ASR actually is.  ASR (Answer Seizure Ratio) is the measurement of successfully answered calls compared to call attempts.  For example, Say you send VoIP Carrier X 100 call attempts and 75 calls are answered.  Your ASR is now at 75%.

But why are the other 25% of calls failing? Busy signals, switch congestion, inaccurate data, inactive dialed numbers, poor network performance and other call errors count as call failures.  Some of these failures such as Busy Signals are a regular and normal part of how telephony functions.  Other failures such as switch congestion and network performance are something that your Network Admin should take a deeper look into.

Here at Star Communication we offer reliable, redundant solutions built for VoIP Carriers, CLECs, Large Call Centers, SMBs and more.  We’ve helped 100’s of our customers improve their ASR with 3 simple inexpensive solutions.


1.) Adding Capacity– Sometimes your underlying VoIP carrier limits the number of channels assigned to your account.  This can cause poor connectivity and in turn create a low ASR.  Starcom gives you instant access to 120,000+ channels at your fingertips.  If you need to urgently get out 1 million + calls an hour, you want to be sure your VoIP carrier has the capacity needed.  With our competitive NPANXX Rate or Flat Rate offers we’ve helped many customers cut cost as well as improve ASR.


2.) Route Comprised of Tier 1 Carriers– Some VoIP carriers have 50+ underlying carriers themselves.  What if your carrier has their 50 underlying carriers in LCR (Least Cost Routing)?  This can certainly create congestion at the switch, a high PDD (Post Dial Delay) and negatively effect your ASR.  Starcom’s underlying network is comprised of multiple Tier 1 carriers.  We help you terminate your calls faster, improve call quality and help achieve a higher ASR.


3.) Caller ID Name– Your Caller ID Name matters when calling customers. Using a local phone number as your Caller ID is usually the preferred method but often your Caller ID Name shows up as “unknown” or “unavailable”.  Most call recipients are not comfortable answering an “unknown” Caller ID.  This can greatly affect your ASR in a negative way.  Starcom has the ability to assign a Caller ID Name of your choice to our local phone numbers.  An accurate Caller ID Name can improve your ASR by 25% or more!  Call Centers, Collection Companies, Non-profit Organizations and other high volume outbound calling customers can benefit and even profit from our Caller ID Name program.  Best of all, this program is 100% FREE with unlimited channels included.


Want to setup a test environment?  Starcom gives FREE testing on all services mentioned.

Complete our contact form here to get started today

sales@starcompartners.com

1-877-519-8464

www.starcompartners.com

25 cent DIDs when you order 25 channels or more

Starcom offers local DIDs (phone numbers) for inbound calling in over 20,000 rate centers and localities in the United States and Canada.  Regardless of your requirements easily establish a local presence in the markets you serve.  Our origination services are designed to provide carriers, re-sellers, and application developers with the proper tools for success.

Order 25 channels or more and get 25 Cent DID numbers!

(channels are $5 monthly, minimum 25 channels, unlimited minutes) 

Get started here today with a custom quote for your needs.

  • Access our Class 5 switch for real time CDRs
  • Ability to adjust peer details
  • Make payments in real time
  • Supports most codecs
  • Built-in high-performance IP firewall
  • Rapid response support
  • Provisioning support
  • Interactive training sessions for all new customers

sales@starcompartners.com

1-877-519-8464

Completely FREE USA SIP Trunk

Bring your company’s phone system out of the Stone Age with Star Communication!

Make calls to any Toll Free number in the USA and Canada for free with no registration required.  1844, 1855, 1866, 1877, 1888 and 1800

100% FREE with CLI delivered

Dial these numbers in international format (+1800xxxxxxx, +1844xxxxxxx, +1855xxxxxxx, +1866xxxxxxx, +1877xxxxxxx, +1888xxxxxxx), and enjoy the call quality!

Trunk details provided here

Format: 18YYXXXXXXX
DTMF: RFC2833
Faxing: T.38 or passthrough (attempt)
SIP Server: switch.starcompartners.com (173.193.144.207)
Port: 5060

Did you set up RasPBX with the latest image? All images from 2017 and later ship with our FREE default trunk included, provided by Star Communication. Check out more on our partnership here

  • No Registration required
  • Tier 1 route
  • CLI Route
  • Unlimited Usage

www.starcompartners.com Be sure to check out out other services!

 

Star Communication Partners with RasPBX

Star Communication is proud to announce a new partnership with the famous RasPBX.  RasPBX is an open-source project that maintains a complete install of Asterisk and FreePBX for the industry changing Raspberry Pi.  Check their download page for the latest RasPBX image.

Did you set up RasPBX with the latest image?  All newer images from 2017 and later ship with a default trunk included, provided by Star Communication.  Use the default Starcom SIP trunk to dial toll-free numbers in the U.S. for free.  Use your SIP capable phone and connect to this extension.  You are ready to call U.S. based toll-free numbers in international format, using the +1 prefix (+1800xxxxxxx, +1844xxxxxxx, +1855xxxxxxx, +1866xxxxxxx, +1877xxxxxxx, +1888xxxxxxx).  Visit here for detailed information.

If you enjoy the call quality, be sure to check out other VoIP services Star Communication offers.

 

  • Want to find out more about RasPBX?  Visit the FAQs page at RasPBX’s website.  
  • RasPBX users with existing setups can easily add trunk and route, details provided here.
  • Please give time to thank the open-source RasPBX project with your donation