Is Channel Capacity placing your VoIP on hold?

Let’s say it’s a busy day in your office and every employee is on the phone making and taking calls. Then another customer needs help and calls your office, except this call is not connected. The customer gets a busy signal..

Yes, even today many VoIP Carriers and VoIP Providers use the same busy signal when a user is at max channels (phone lines). Customers expect phone calls to connect when calling a business. Even if calling into an IVR and having to go through a short call queue, no one expects to hear a busy signal. This can look bad for your business overall.

Whether you’re a VoIP Newbie or have 10 certifications, it’s a good idea to review any VoIP provider’s plan and channel limitations they offer. Many of the packages and pricing models from providers do indeed have a limited number of channels. Another thing to help prevent customers receiving a busy signal, is to plan with your VoIP provider around your peak calling seasons. Maybe your business’s call volume is highest around Christmas. Ask your VoIP provider ahead of time to increase the number a channels if you expect a heavier call flow.

Star Communication offers SIP termination and Origination that is reliable, redundant and built your business. If you need to urgently make or receive 1 million calls an hour, we give you instant access to 120,000+ channels and more than 8,500 CPS at your fingertips. Typically we are able to offer better rates and assist our customers in choosing the right solution. We have helped VoIP carriers, CLECs, Large Call Centers and we look forward to assisting you!

www.starcompartners.com
Phone – 1(877) 519-8464
Email – sales@starcompartners.com
Skype – andrew.helton50

Recent Rate Update

Star Communication has recently rolled out their rate deck builder to better enhance performance and rates.   All of the work was done based on our customers’ important feedback.  Unlike the industry standard practice, Starcom rate decks will be customized to take into account each customer’s specific needs and requirements.  Including but not limited to the type of traffic, capacity needs and more. The recent changes will translate into the following positive impact-

  • Enhanced routes resulting in shorter PDD
  • Major service improvement
  • Carrier selection optimization
  • Ensuring customers always have access to the most competitive rates

Feel free to check out our offer rates and reach out directly here

Star Communication offers SIP termination that is reliable, redundant and built for the highest call quality . Typically we are able to offer better rates and assist our customers in choosing the right solution. We have helped VoIP carriers, CLECs, Large Call Centers and we look forward to assisting you!

www.starcompartners.com

Phone – 1(877) 519-8464
Email – andrew@starcompartners.com
Skype – andrew.helton50

What is ALOC in VoIP?

ALOC is an acronym that stands for Average Length of Call.   ALOC is a measurement determined by looking at a specific time frame of your VoIP traffic and finding the average call length.  Most often your CDR (call detail record) or other reporting tool will give you this report.  ALOC is used to monitor traffic, measure call quality, find the type of call traffic, predict network demand and more.

The acronym is ACD (average call duration) has virtually the same meaning as ALOC.  However, ACD will also be used to represent Automatic Call Distributors in telecommunications.  Which is un-related to this topic.

Telephone calls vary in the duration of minutes used.  Calls could be as short as 6 seconds or longer than 1 hour.  If you were to compare the ALOC of a Call Center vs. traditional outbound calling, the Center would have a much lower ALOC.  Call Centers make more call attempts than your average business or home owner.  Many outbound calls that Centers make go un-answered by the customer or are very short in length.  Call Centers also have an obligation to handle inbound calls.  Their first priority is to help resolve the customer’s issue and second is to move onto the next caller that is waiting.  Call centers also use advanced outbound dialers and call routing software to enhance their call capacity.  This typically results in an ALOC under 1 minute and a high demand on network infastrcuture.  Traditional business or homeowners are making calls 1×1 and dialing by using their hands.  Thus their ALOC is often greater than 2 minutes and there is less of a demand on network infrastructure.


Star Communication’s voice platform was built by a team of telecommunications experts and has effectively dialed billions of calls for its clients during their critical communication times. Whether you’re a VoIP Carrier, CLEC, Call Center or other VoIP operator rest assured we will meet your demand with quality and reliability.

For rates and more information on our services, visit our website here

Phone: 1(877) 519-8464
Email: sales@starcompartners.com

Wholesale VoIP Carrier Solution

Star Communication offers a completely managed and customizable Wholesale VoIP carrier solution. We provide a fully-redundant carrier-grade termination service on one of the industry’s most extensive voice networks. Starcom allows you to take full advantage of the best available call quality and cutting-edge solutions. Our, custom-built, least cost routing engine enables us to route calls strategically, giving you the best possible pricing. We’ve also created strategic relationships with top telecommunications carriers, leveraging our high-volume traffic to negotiate aggressive rates.

What does that mean for you? Nothing less than the best possible wholesale pricing in the industry.

Of course we’d love to help you cut cost on your destinations. You’re able to find current rates listed on our website here

 

Phone: 1(877) 519-8464
Email: sales@starcompartners.com

What is a CDR?

CDR is a VoIP acronym that stands for Call Detail Record.  A Call Detail Record is a comprehensive file of a recently completed call or call attempt over your VoIP network.  The information most often contains a Time and Date stamp, the CLI (caller) phone number, the CLD (callee) phone number, country of origin, call duration, billed call duration and charged amount.  Typically CDRs will be included with your monthly phone bill much like your traditional telephone provider.

More advanced VoIP applications and VoIP Carriers require highly detailed CDRs.  Call-ID, Delay, PDD, User Agent, and LRN information are just a few items included on highly detailed CDRs.  Some VoIP applications also require real-time CRDs from their VoIP Provider.  Most often this is offered via an online user portal.

Starcom CDRs Include-

Parameter Value
Call-Id
CLI
CLD
Protocol
I-Protocol
I-Cdr
I-Call
I-Account
Result
Cost
Delay
Duration
Billed Duration
Connect Time
Disconnect Time
Incoming CLD
Incoming CLI
Prefix
Price 1
Price N
Interval 1
Interval N
Post Call Surcharge
Connect Fee
Free Seconds
Remote IP
Grace Period
User Agent
PDD
Release Source
Plan Duration
Accessibility Cost
LRN CLD
Incoming LRN CLD
Area Name
P-Asserted-Id
Remote-Party-Id
Connect Processing Time


Need Long Distance SIP Trunking?  We offer simple Outbound SIP Termination for your PBX at $.009/minute with all of US and Canada included.  Inbound DID rates are found here and Inbound Toll Free rates are listed here.

Carrier accounts may download our wholesale NPANXX deck here.  

Phone: 1(877) 519-8464
Email: sales@starcompartners.com

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

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

 

Low rates for high quality USA CC VoIP route. As low as $.0045

Star Communication has a special pricing model that allows us to provide the highest quality short duration termination.   We service VoIP providers, CLECs, PBX providers, SMBs and contact centers around the globe.  With competitive pricing and reliability, we’ve helped many VoIP carriers and call centers cut telecommunications cost.

Customers using our full NPANXX rate plan often average costs as low as $.0045 per minute.  Most Call Centers call highly populated areas which tend to be lower in cost per minute.  For example; a center is more likely to call San Antonio, TX [rate $.0019/min] than Fairbanks, Alaska [rate $.02/min].  Prices vary because phone companies have different billing arraignments and hard costs to route and terminate calls to their subscribers.   Highly populated areas are more competitive in pricing which drives VoIP termination costs lower.  Much of Star Communication’s coverage starts at just $.001 per minute.

Curious how much we could save you?  Find out with a FREE test today!

Fill our test form here and get a test account in as little as 2 hours

  • High CPS
  • High Capacity
  • Tier 1 Direct Interconnects
  • Web-access to monitor traffic
  • Easy online payments
  • Free Testing
  • Short-duration Traffic welcome

 

Contact – 1(877)519-8464 Ext. 800

Email – andrew@starcompartners.com

Skype – andrew.helton50

Website – http://starcompartners.com/

Carpe diem

5 simple ways to cut VoIP expense in 2017

Looking to cut down telecommunication costs moving into 2017? Here are a 5 simple ways to add value and cut expense on your VoIP infrastructure-

1.) Localized Caller ID Numbers in 48 states

When you call your customers in 2017 let them know who is calling with localized DIDs with Caller ID data. There is NO cost to you for DIDs or inbound minutes.  CNAM dip commissions 50% total collected.   Visit here for more info

2.) FREE Toll-Free termination

Stop paying for those long outbound calls to 1800, 1888, 1877, 1866, 1855, 1844 (most carriers charge a high rate for these calls).  We offer FREE redundant SIP termination to all USA & Canada Toll Free Numbers.  Visit here for more info

3.) Local DIDs and Toll Free numbers 

Local numbers (DIDs) 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.  Visit here for more info

4.) Wholesale USA & Canada SIP Termination

We have a special pricing model that allows us to provide the highest quality wholesale termination at very competitive rates. 20,000 channels available with full coverage.  Great for your In-house PBX, Hosted PBX providers and Wholesale VoIP Carriers.  Visit here for rates

5.) Wholesale USA & Canada Dialer termination

We will deliver service that allows customers that use auto dialers to take advantage of new technologies while simultaneously keeping costs low.  35,000 channels available with full coverage.  Our wholesale rate decks are designed for high volume call centers, VoIP Carriers, and least cost routing engines.  Visit here for rates

*Note #1 and #2 are completely free
Star Communication offers free testing on any of these services. If you’d like to discuss the value of these products in more detail, feel free to shoot us an email at sales@starcompartners.com or call 1-877-519-8464

 

Star Communication COO- Andrew Helton

Linked In – http://goo.gl/VMzC1r
Skype – andrew.helton50
www.starcompartners.com

Carpe diem

 

Free VoIP Calling to Business Phones

What if your calls and faxes to businesses we completely FREE? Call Centers, VoIP Providers, and savvy business owners are taking advantage of this VoIP technology. Star Communications welcomes companies to significantly cut telecommunication cost with this simple application.

In the search for any business’s phone number, 8 times out of 10 the business will have a Toll Free number listed. When you call a Toll Free number over the Starcom’s SIP Trunk, we guarantee to route and terminate those calls with no minute charges to you. You heard this right, completely FREE. This would apply to conversational as well as fax traffic to the 1800, 1833, 1844, 1855, 1866, 1877 or 1888 prefixes.

Below are the Top 3 ways a Toll Free termination provider can improve your bottom line-

  1. Check with your current provider, you may be getting charged for this traffic. Potentially we could save you money starting today.

  2. If your business has only a certain amount of ports available. Sending this traffic to a Toll Free termination provider would free up additional ports on your network.

  3. Highest quality sound and ASR. Bypassing traditional routes your call is delivered near direct to the carrier who owns the 18YY number. Cutting out many routes in-between and providing you with the highest possible quality of service. 100% CLI delivery with all calls.

Toll Free termination can apply to anyone who operates a PBX or Telephony switch. VoIP is an ever expanding technology and is being utilized in ways it has never before. New technologies are constantly being developed and companies are finding those easy ways to cut or eliminate cost.

Visit here for instructions. Or see the below configuration

Format: 18xxxxxxxxx
DTMF: RFC2833
Faxing: T.38 or pass-through
SIP Port: 5060
SIP Server(s): ovh.starcompartners.com (144.217.68.110)Montreal, QC switch.starcompartners.com (173.193.144.207)Dallas, TX
Questions? Click here to complete our simple contact form. Someone will reach out in a timely fashion.