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

Spring into Savings on VoIP Termination

Greetings from Star Communication, 

We are pleased to announce our $0.0034\minute flat USA that is winning quite a bit of traffic.  Additionally our Toll Free Termination product has increased in size adding our 3rd location in Hamburg, Germany.  If you would like to test or have any questions about any of our products below, reach out to us and someone will assist you with testing.

1. USA Retail Conversational $0.0034\minute 6\6 billing.  No Dialers
2. Free Toll Free Termination (No account needed).  Complete your 18XX traffic with 100% direct to tandem access.  High volume we can discuss revenue sharing agreements
3. USA NPANXX CC\Dialer.  Customers average $0.003
4. Australia Proper $0.02.  Mobile $0.038. 1\1 Billing
5. Hosted Broadcast IVR as low as $0.0076\minute.  6\6 Billing. 30,000+ ports available

For free testing on any of these products please reach us at-

www.starcompartners.com

Phone: 1(877) 519-8464
Email: sales@starcompartners.com
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Fill your SIP Trunk with Goodies from Starcom

Local DIDs just $.25 cents each

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.  


Toll-free Numbers 

Minutes low as $0.007 per.  Free TFN Ordering & Porting  


SIP Termination with 99.999% up time

We offer SIP termination that is reliable, redundant and built for VoIP Carriers.  If you need to urgently send millions of calls an hour, we give you instant access to 120,000+ channels at your fingertips.  Typically we are able to offer better rates and assist our customers in choosing the right solution.  


Ringless Voicemail

Earn your Black Belt in Marketing with our Ringless Voicemail Ninja.  Rates start at $.015 per successful drop.   Get 50 Free credits when you register today!


Hosted Voice Broadcast for just $.009/minute

Send interactive voice messages or reminders to thousands of contacts per minute.  Build interactive voice functions using recordings, dynamic TTS, and speech recognition.  You can always get setup same day and typically within the hour.


Hosted Bulk SMS

Send and receive BULK SMS messages with long code, short code, and toll-free telephone numbers.


Hosted IP PBX

Combine our inexpensive Long Distance calling and DIDs with our Zero cost IP PBX hosting!    Get started here today with a custom quote for your needs


For free testing on any of these products please complete our contact form here

Or email sales@starcompartners.com 

Happy Holidays from the entire Starcom Team

 

 

Take a “Peak” at Starcom’s Summer Specials!

Thanks for taking a “peak” at some of our Summer Specials!  If you would like to test anything on the list please feel free to reach out to sales@starcompartners.com directly.

1) USA Premium Dialer.  NPANXX Deck Rates.  6\6 Billing.  Averaging Below $0.005\minute.  Rates here


2) USA\Canada Hosted Voice Broadcasting $0.0076\minute.  6\6 Billing. 36,000+ ports available.  Press 1, IVR, Ringless Voicemail, Speech Enabled


3) Canada Flat $0.0031\minute.  6\6 Billing


4) 100% Free Toll Free(18YY) Termination.  Complete your companies Toll Free traffic over 100% direct to tandem routes.  Unlimited Ports.  Format here


5) UK Dialer Proper $0.0035.  Mobile $0.0145.  1\1 Billing


6) Australia Proper $0.02.  Mobile $0.038.  1\1 Billing


7) USA Caller IDs.  Save up to 10% on your USA Dialer campaigns without changing carriers.  Improve your ASR with local Caller IDs.  More info here


8) China Open RTP PCCW $0.062\minute.  China Closed RTP TATA $0.057\minute. 1\1 Billing.  NO MISSED CALLS


9) India All Flat $0.015\minute.  1\1 Billing

 

For free testing on any of these products please complete our contact form here

Or 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