HELP! - Phones only works half the day

Asterisk using FreePBX with Astra 53i phones. Asterisk and FreePBX is hosted on the cloud from Storm on Demand (Liquid Web).

I would love anyone’s help with this. I appreciate the help from this forum.

So my phones only work half the day. Here is what is happening when my phone isn’t working…

I would dial out, the I don’t hear it ringing through my phone. I tried calling my cell phone, and the call does go through, but most of the time (not all the time), when I dial out, the other side cannot hear me. I can hear them, but they can’t hear me.

during this time, when you try to call my phone, if I don’t answer, it’s fine, cause it goes to voicemail and I get the vm, but if I pick it up, it terminates the call on the other side.

Like I said, it seems to happen about half of the day. The times differ, and seems to be getting later and later. Used to be at about noon, it would be ok.

I can tell when this issue is occuring, cause I just dial my cell phone, and if I don’t hear the ringing on my astra phone, then the connection is bad.

I tried changing phones, to see if it was the hardware, but same thing with other phones.

I have a consultant that’s been trying to help me, but no luck.

What we are doing right now, is we are changing sip trunk providers, and in the meantime, we have 2 sip trunk providers opened on our freepbx. It’s taking much longer than I would have liked, cause my previous provider is rejecting the porting requests, but it should be getting taken care of.

Please help if anyone has experienced this before, or knows what this could be, as I am lost, and I can’t work this way.

Thank you.

Definitely network, not Asterisk.

Does your ISP provide a well behaved IP address (does not change whilst the xDSL is up, or across short down times)?

I haven’t had time with my network before. My asterisk is sitting on a dedicated server running Centos with 2gb ram. It hasn’t caused problems before, and the load on that server is minimal.

However, my consultant said there is a latency of around 300ms from the phone provider into our network. is that enough to cause what is happening?

The server is running on a dedicated IP, and i have only been having this issue for a month or so now, and not before.

An each way latency of 300ms is likely to make VoIP quality unacceptable to anyone except astronauts on the moon, or beyond. It’s about an order of magnitude too high for VoIP use by the general public.

If you have qualify set to yes, Asterisk will tolerate round trip times up to 2 seconds. If you have it set to a number, you will need to check the configuration. In any case, you will get entries in the Asterisk logs if this limit is exceeded. If you have qualify off, you will get retransmissions after 500ms round trip time, but these will not harm a compliant SIP UAS.

I would suggest that you look at the logs from the Asterisk server.

David,
thank you for your help and attention to this.

I have been looking into the network of my host, and my isp here where my sip phone is actually being used. The networks say that there isn’t enough of a latency to cause voip service to be effected, but we are still seeing 400 ms latency from host to the physical phone. I will research this a bit more, and post when I find out.

Again, i appreciate your help in this.