Horsing around with your VoIP Provider?

Horsing around: To engage in aimless recreation or frivolous time-wasting; to fool around.   Retrieved from –https://idioms.thefreedictionary.com/horsing+around

Maybe you are looking for a new VoIP  interconnection and the process is taking too long.  Or you have a new IP that needs whitelisted by your carrier immediately.  Maybe there is a trouble number that is not connecting.  Perhaps the call quality is suspect and reliability is wavering.  Hopefully this is not your experience with VoIP.  But if it is, not all VoIP carriers are created equally and Starcom in not your old horse and buggy VoIP Carrier.

Who doesn’t like an an easy overview?  Here are a few pointers of our wholesale telecom service-

  • Rapid response support (Skype, Phone and Email)
  • Access our carrier grade Class 4 switch for CDRs
  • Self manage Source IPs
  • Make payments in real time
  • Support most codecs
  • Built-in high-performance IP firewall
  • 120,000 channel capacity
  • Extremely low ping times
  • Provisioning support
  • Interactive training sessions for all new customers

We appreciate your interest in Star Communication. Starcom offers flexible wholesale US and Canada VoIP termination and origination.  Our services are built for the highest call quality direct to our Tier 1 underlying carrier.   Interconnected with hundreds of customers and carriers, we have helped Large VoIP carriers, CLECs, PBX Providers, Call Centers and more cut costs on VoIP Termination.

For more information, visit us at- www.starcompartners.com
Phone – 1(877) 519-8464
Email – sales@starcompartners.com

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

Star Communication Announces Partnership with iNextrix and Integration with ASTPP

iNextrix Technologies Pvt. ltd., a next generation IT company and project leader of ASTPP, an Open Source VoIP Billing application for Freeswitch® & Star Communication, a leading global IP based service provider of SIP based
local, toll free and long distance services, today announced a technology partnership and integration agreement.

The partnership provides users of ASTPP with access to Star Communication SIP trunking services, rates and DID configuration from within ASTPP.  The partnership also provides Star Communication customers with direct access to iNextrix services and support.

 
“We are glad announcing our partnership with Star Communication”, said iNextrix’s Director Samir Doshi. “This partnership will make great benefits to everyone, specially for the companies who are starting VOIP business and need a reliable open source VoIP billing platform along with a good termination provider. This integration is bringing all together and make it ready to go in a single shot.”, said Samir.

About iNextrix


iNextrix Technologies was founded in 2010 by two young and dynamic entrepreneurs Mr. Arpit Modi and Mr. Samir Doshi with the goal to provide the best service and development in industry at a competitive rate with 100% client satisfaction.  Initially, the company was mainly working on VoIP based technologies and later in 2011, Inextrix expand their wings and bring web and mobility development under his umbrella whereas started building team of talented and innovative resources.

With years of experience in Open Source softswitch platforms such as: Asterisk®, Freeswitch®, Opensips® and Kamailio®, Inextrix uses the knowledge gained to build software applications to enhance these platforms and assist Service Providers in offering carrier grade VoIP services built on open source standards.  Their core platform is ASTPP, which is an Open Source VoIP Billing application for Freeswitch. Inextrix provides commercial installations, configurations, support as well as custom applications development. For more information, visit http://www.inextrix.com.

About Star Communication

Star Communication offers SIP Termination and Origination that is reliable, redundant and built for VoIP Carriers.  We have helped VoIP carriers, CLECs, large Call Centers and most of all we look forward to assisting you!

Based in the USA, Starcom’s voice platform has effectively routed billions of calls for its clients during their critical communication times. We specialize in SIP to Tandem and Wholesale VoIP Termination for the US and Canada.  We also offer many wholesale Tier 1 routes to A-Z countries for customer calling internationally.  Companies across the globe look to us for telecom savings and we invite you to see why!

Services-

  1. Wholesale VoIP Termination
  2. SIP to Tandem Tier 1 Termination
  3. Short Duration Termination
  4. A-Z Termination
  5. Free Toll Free Termination
  6. VoIP Origination
  7. DIDs and Toll-Free Numbers

For more information, visit https://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 Ways to Improve ASR on 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 XYZ 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 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.) Toll-free Termination- Using our Direct to Tandem route, calls placed to United States and Canadian Toll-free numbers are 100% Free. We often are able to help users improve their ASR along with PDD. We offer unlimited usage and CLI delivery.

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


Want to setup a test environment?  Starcom gives FREE testing and interactive training for all new customers,

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

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.