Assessing An Existing System What Do I Have How to Get Help

Hello Forum,

I have been contacted by a new client who has in place an Asterisk system and is having issues with dropped calls and distorted phone conversations. I know very little about this product. It would seem that the system was installed by a ‘friend’ and that friend no longer has the time to help.

Can anyone help me to understand what information I would need to gather to start the process of assessing the system. It’s a small real estate business.

One two specific questions:

Can a report be run from the system to provide an idea of what hardware and software is in use?
Can a report be run from the system to provide a history of dropped calls or other problems or errors?

Thank you in advance for any help you can provide to get me started on helping determine what I am dealing with.

The best approch for you is to redirect your client to a vendor with a good asterisk knowledge. In the other hand if you want to start offering asterisk consultancy then you have good alternatives to do it:

About your questions:

  1. Linux has specific commands to retrieve hardware information like hdparm, top, free or cat /proc/cpuinfo.

  2. Asterisk has a logger utility, then you can chek the logs to match the callerid and the hour that happened to try to troubleshoot. Maybe you will need more intense debug depending on the technology used.

Dropped calls issue depends on the technology used by the system, distorted calls are usually due to bandwidth issues.

1 Like

You could also check if the customer is using some Asterisk GUI, Like FreePBX, or Elastix, and If so all the administration is managed through a Web GUI. Related to dropped calls on SIP devices this problem most of the time is due to NAT issue, Network problems , If using Dahdi this could be due to : physical layer or data link layer failure at the remote party or user equipment .

Asterisk or any existing GUI does not provides a report about dropped calls, you need to start doing a SIP trace if using SIP…

I suggest you first investigate the technology used, the network environment, Asterisk version and if any GUI present.

1 Like

Let me start by thanking you for your prompt reply - much appreciated! - It’s good advise to bring in an expert to help this client. I am trying to gather a bit of information on the clients setup prior to making that recommendation.

  • Nature of the issues experiences
  • Hardware and Software in place
  • Reviewing billing to see what service they currently have from broadband providers

I do not currently have a referral from a trusted colleague. I am reaching out to a few vendors I use for non-Asterisk solutions.

This may be a familiar story to you - a friend set it up and is no longer able to manage the system for this client of mine. It a new client for me working with them on social media and web design. They asked for me to help them find out the truth about what is happening.

Again thank you very much!

Thank you @ambiorixg12,

I have been reading that’s how I found this forum - appreciate the quick replies and advice!

I am attempting to interpret the billing from in this case Windstream - I expected to see a billing line for SIP trunking.

I see 3 charges Internet related but nothing indicating the purchase of SIP services. Pardon my ignorance however SIP would be required in-order to use an Asterisk system correct?

If you would also - can you tell me does an Asterisk system have the ability to reserve bandwidth to prevent others using the in this case 12m/1m bandwidth from being consumed by Ethernet connected computers and depriving the PBX of the needed bandwidth?

Thank you in advance for your replies!


There is no requirement for SIP in an Asterisk system. Asterisk was initially developed for analogue and circuit switched systems. There is no requirement for SIP in a VoIP Asterisk system, either e.g. you can use the Cisco native protocol, or IAX.

Prioritisation of traffic is the responsibility of the router, and to some extent the OS. For SIP, Asterisk allows type of service codes to be set on packets, to allow routers to recognize the packets, but it is also possible to do this based on just the port numbers, in some cases.

1 Like

Thank you @david551,

I spent sometime with Windstream and several of their departments determining what services are in play. I have the clients Windstream bill and their Time Warner bill. We have determined that the SIP trunks are likely in use but not being sold by Windstream. So I need to get that bill from the client to see how many channels they have purchased.

I am going to do as good of an inventory of hardware, software and connectivity as possible onsite then likely do a few calculations once I get a good guess on the number of inbound and outbound calls. This will help us to determine if their bandwidth is in the ballpark. Then next I will seek out an Asterisk expert to take over the task of working with the PBX, SIP and Internet Providers. I will have a network diagram with details to provide to them.

I am the city of North Ridgeville in the State of Ohio, If anyone knows of someone they can refer that would great!