DIDForSale Customer Satisfaction survey closed

We have closed the customer satisfaction server. We really appreciate the time and effort you took to complete the survey. As always your input is very important to us. We are putting the reports together and will be sharing with you in a day or two.

Best VoIP Service Providers

www.didforsale.com

Virtual Number for calling card

Over the last few years calling card business have been very competitive. Customers have too many options. Companies have to pay for origination, termination, manage the systems networks. With all these expenses it has become very competitive that directly affect your profits. We have noticed the termination cost going down quite a bit over last few years and most of the companies get good deal on termination. But still lots of people are paying too much for origination.

Some of our customers switched to didforsale and saving lots of money every month. We offer Inbound Virtual numbers specially designed for calling card companies and call centers. Our flat rate inbound DID are just $8.99 per month. Each DID comes with 20 channels where 20 people can call at the same time. Normally a good calling card company takes 50 to 300 numbers and uses 400 to 2000 simultaneous channels. So lets take an average of 150 numbers and 600 channels.

Your cost at DIDforsale will be $8 x 150 = $1200 + tax per month. You don’t have to pay anything extra, no per minute charges and no per channels costs. While other VoIP service providers charges very low cost on the DID but charges very high price on per channel cost. Based on our research average market price is $1 per DID and $12 per channels. So the same company with 150 DID and 600 channels will be paying:
150 x $1 + $12 x 600 = $7350 per month.

Compare the prices $1200 vs $7350.

Talk to our sales professional and let us help you save money on your VoIP needs.

VoIP Service Provider
www.didforsale.com

Troubleshooting SIP Service with ngrep

If you are using VoIP (Voice over IP) then at some point you must have to troubleshoot the call flow to find where the problem is. There are many tools you can use. To teach you exactly how to use a protocol analyzer is beyond the scope of this material; however, we will give you some tips on analyzing SIP and RTP packets. Bear in mind these tools can do much much more than what I illustrate here. I will be using these tool just to capture VoIP traffic.

Most common used tool are: Ngrep, Wireshark, TCPdump.

Ngrep:
We often use ngrep to monitor our VoIP services, because it is very simple and light. If you don’t have it installed, then you can download it from this link
http://yum.trixbox.org/centos/5/old/repodata/repoview/ngrep-0-1.45-1.el5.rf.html
http://yum.trixbox.org/centos/5/old/ngrep-1.45-1.el5.rf.i386.rpm

This command will capture everything on SIP port (I know this can capture any network activity, but I am using it for VoIP on SIP ) and dump the data on the screen. Easiest way to know how your system in interacting with your vendor or client.
See everything on your system on SIP protocol.

ngrep port 5060

If you want to filter the traffic based on phone number
ngrep ‘9498859944’ port 5060

If you want to filter traffic based on IP address
ngrep ‘IPAddress’ port 5060
Example
ngrep ‘209.201.2.255’ port 5060

Other important options
-t     Print a timestamp in the form of YYYY/MM/DD
-T     Print a timestamp in the form of +S.UUUUUU, indicating the delta between packet matches.
Example
ngrep  -t ‘9498859944’ port 5060
ngrep  -T ‘9498859944’ port 5060

As I said there are lot more options read man pages ‘man ngrep’, but this is good enough to see the call flow on your system.

VoIP Provider,
www.didforsale.com

What is SIP TRUNKING?

SIP is a Session Initiation Protocol at application layer which controls creating, modifying, and terminating sessions with multiple participants. A SIP Trunk can be referred to as a logical connection between an IP PBX and a Service Provider’s application servers that allows voice over IP traffic to be exchanged between the two.  While placing a call PBX routes the information to the SIP Trunk provider who establishes the call to the dialed number and acts as an agent for the call. All signaling and voice traffic between the PBX and the provider is exchanged using SIP and RTP protocol packets over the IP network. To successfully deploy SIP trunk following three components are essential
  • PBX with a SIP-enabled trunk side,
  • An enterprise edge device understanding SIP
  • Internet telephony or SIP trunking service provider

Industry Survey: SIP Trunking in the Enterprise is Poised for Growth in 2011…But Significant Operational Questions Remain

SIP Trunking – What does the future look like?

A new independent survey research report on Session Internet Protocol (SIP) trunking reveals an enterprise marketplace that is poised to make significant investments in this technology – even as executives wrestle with questions about exactly how new deployments will affect current telecom operations and future initiatives.

The survey of 138 executives with telecommunications decision-making responsibilities was conducted and published by Voice Report, a publication of CCMI, in conjunction with BizTechReports, an independent reporting agency that covers enterprise technology trends.

To read complete story click here