Experiencing choppy calls both extension to extension and outbound call

Hello community,

recently i bought a dedicated server, bellow is my server specification;

processor : 0
vendor_id : GenuineIntel
cpu family : 6
model : 63
model name : Intel® Xeon® CPU E5-2650L v3 @ 1.80GHz
stepping : 2
microcode : 0x1
cpu MHz : 1799.998
cache size : 30720 KB

MemTotal: 500836 kB

I’ve asterisk-13.6.0, after a successful setup everything seemed to be ok not after when i created two extensions and tried to call another extension, i get a choppy/broken voice, I don’t think it’s my network cos i tried different softphones on different networks, changed different Codec(s), reinstalled but the issue is still the same. I don’t know what to do next, is it my server which is very weak OR? please advice me

checking in…No reply!
please guys help me out

Your 24 hours minimum bump time isn’t up yet.

You appear to have excluded the only likely cause.

thanks David! if i may ask, what likely other causes, (because i configured asterisk on my personal PC and everything works perfectly…) I don’t know what’s happening with this one
I’ll be grateful if you do pinpoint some of the causes…

Running on a virtual machine. Running CPU intensive applications on the same machine.

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hello david, seriously i don’t get what you said, please could you try to explain it to me.
you mean virtual machines can’t get me what i want?

Virtual machines need very careful configuration to achieve under 20ms scheduling latencies and accurate internal times, as they were not, originally, designed for real time applications, like telephony.

Hello David, thanks for that little explanation…I’ll only ask you one thing! how would you advice me?what do i need to do inorder to achieve this?
Well, my main aim is to startup a small voip business in my country(uganda), so i need this please. I know how to install/configure asterisk but maybe am missing that little thing. Can you please refer me, somewhere i can read more about this.

Hello David!
I want to thank you for your time and advice good of it, i found out what the problem was. so i just wanted to update it maybe it’ll be useful to someone in the coming time.

Well, i didn’t mention that i was using Digital Ocean servers, so here is the warning I got from someone who was using the same server:

WARNING: If you’re using a 512MB droplet at Digital Ocean, be advised that their setups do NOT include a swap file. This may cause serious problems when you run out of RAM. Uncomment ./create-swapfile-DO line below to create a 1GB swap file which will be activated whenever you exceed 90% RAM usage on Digital Ocean.

Immediately after creating the swap file (i gave it 4GB) every is now normal.

But their is only one issue now am facing. if i use a phone supporting 3G+ network to make a call to another, everything is good and clear, but if i use a phone supporting only Edge network, the calls are not clear at all.

how do I make my pbx to support edge network too just like 3G network.

Edge is unlikely to have enough bandwidth to complete a VoIP call unless using a compressed codec like G729A. As well the latency will make the quality of the call very poor as well.

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