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

Starcom adds Toll-free Termination POP in Los Angeles

Starcom’s Toll-free Termination

We are excited to announce our new Los Angeles, California POP for our Toll-free Termination partners!  Our new POP in L.A. allows us to strategically service our partners based on the West Coast and in Asia.

Our Dallas and Montreal POPs are still available of course.  You also may use 2 or more POPs for high volume load balancing or a high-end failover situations.

Want to try it out?  Within your SIP device’s Dial-plan simply add our Toll-free Termination Carrier settings and Star Communication will route traffic with zero costs to you.  No account required.

 

Format: 18xxxxxxxxx
DTMF: RFC2833
Faxing: T.38 or pass-through
SIP Port: 5060
SIP Server(s):  (Choose the lowest latency)

72.11.130.234 Los Angeles, CA
144.217.68.110 Montreal, QC
173.193.144.207 Dallas, TX

Toll-free Termination is a VoIP term related to outbound SIP termination calls placed to United States and Canadian Toll-free numbers. Typically there are no costs to terminate this traffic for the end user.

www.starcompartners.com

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

What is A-Z Termination in VoIP?

A-Z Termination, also known as Afghanistan to Zimbabwe Termination, implies that a VoIP Carrier offers VoIP  Termination routes over the entire globe.  Listing all of the countries in the world, in alphabetical order, Afghanistan comes first and Zimbabwe last.   It makes sense to abbreviate the two.

Other abbreviations such as A-Z cc CLI, A-Z CLI, A-Z NCLI often seem like a mouth full.  They’re just a few relatively simple terms that represent the type of A-Z Termination route the Carrier offers.

A-Z cc CLI stands for A-Z Call Center CLI termination.  The CLI acronym stands for Caller Line Identification.  CLI implies that the Caller ID of the Caller with be portrayed to the call-recipient.  Call Centers and large businesses typically have low  ALOC.  Thus network demand on the Carrier is greater which typically leads to more expensive rates to compensate for the load.  Short duration, Call Center, traffic typically has an ALOC below 1 minute.

A-Z CLI implies the Carrier offers VoIP termination across the globe and will portray the Caller ID of the Caller to the recipient.  A-Z CLI offers will give the lowest PDD (Post Dial Delay) and offer the clearest audio.  Most carriers expect an ALOC above 1 minute over a A-Z CLI route.   A-Z CLI routes are less expensive than A-Z cc CLI.

A-Z NCLI is an offer that has global coverage but does not guarantee the Caller ID will be displayed to the recipient.  Most often this is best attempt at displaying Caller ID and is sometimes referred to as a grey route.  NCLI routes can be the least expensive but are more difficult to manage without CLI.  ALOC requirements for NCLI are negotiable with each carrier.


Looking for A-Z Termination?  Star Communication offers VoIP services across the globe and unlimited capacity for most destinations.  Check out our latest A-Z cc CLI and A-Z CLI rate decks here

Hit our Free Testing button today.  Use $5 in credits to check CDRs, call stats and more.

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

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

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

Starcom adds 3rd Toll-free Termination POP in Germany

Starcom’s Toll-free Termination

We are excited to announce our new Hamburg, Germany POP for our customers based in the EU and Middle East!  Our Dallas and Montreal POPs are still available of course.  You can also use 2 or more POPs for higher volume load balancing or a higher end fail-over situation.

From any where in the world, 100% free, make calls to any 1800, 1833, 1844, 1855, 1866, 1877 and 1888 numbers in the USA and Canada.  No registration required and full CLI delivered

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 germany.starcompartners.com (54.37.91.250) Hamburg

Find more information on our website here


Need Long Distance SIP Trunking?  We offer simple Outbound SIP Termination for your PBX at $.01/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.  


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

 

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

Completely FREE USA SIP Trunk

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

100% FREE: Make calls to any 1800, 1833, 1844, 1855, 1866, 1877 and 1888 numbers in the USA and Canada

  • CLI delivered
  • No Registration required
  • Tier 1 Tandem route
  • Unlimited Usage

Choose from our Montreal(OVH), QC or Dallas, TX locations. You can also use both for higher volume load balancing or a higher end fail-over situation.

Trunk details provided here

Format: 18xxxxxxxxx
DTMF: RFC2833
Faxing: T.38 or pass-through
SIP Port: 5060
SIP Server(s): ovh.starcompartners.com (144.217.68.110) Montreal
switch.starcompartners.com (173.193.144.207) Dallas

Our tier 1 and direct carrier partners provide us the ability to route calls with the up-most priority and redundancy to ensure your customers are happy. We are constantly monitoring our network for jitter, latency, and upsteam carriers outages. Giving our customers piece of mind so they can focus on what matters.

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


Need Long Distance SIP Trunking?  We offer simple Outbound SIP Termination for your PBX at $.01/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.  


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

 

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