attended transfer major problem : phone issue

[ajpoole]

Hi

I'm running 50 phones in a serviced office environment,
3CX is running on a VPS in a remote datacentre, server 2008 r2, 8Gb Ram, all extensions are external.
Router is a cisco 2821, on a dedicated annex m line
Phones are Grandstream GXP2100, they are provisioned/served firmware/served xml phonebooks etc all by 3CX

most issues are ironed out now (I hope) but one remains:

Steps to recreate the problem:
1. make an internal call from ext a
2. answer it on ext b
3. ext a places the call on hold and rings ext c
(ext b gets on hold music)
* ext a and c are stuck in the call i.e. can not hangup and the in-call menu prevents manually rebooting so phone appears froxen (but isn't - just cant hang up)
* ext b just stays on hold.
* 3CX shows two calls which can then be manually disconnected
* the phones have to be physically upplugged to reboot
* 3CX logs show '408 request timed out' for the ext b to c leg of the call
* 3CX doesn't know theres a registration problem but cant reach the phones

I then have to reboot all of the phones to even get dial tones!

This problem seems to be new and related to the audio streams no longer going through the server ('3CX: pbx delivers audio' = false) which I changed yesterday, I think an internal transfer is a direct IP call, so I checked the phone config and its not disabled.

I have attached the 3CX log covering the time of the call


Please help if anyones experienced anything like it, every time this happens it kills the entire system!

Thanks

Andrew

 

[ajlinggi]

Re: attended transfer major problem

Hello,

I had the same problems... amongst others with the GXP2110.

Ended up having to replace the phones for the client at my own cost with Yealink T26 and T28's

Spoke to Grandstream about it yesterday and they reckon that beta 107 firmware fixes all the issues. I tried it and it didnt.

Maybe you'll have more luck.

Cheers,
Adrian.

 

[ajpoole]

Re: attended transfer major problem :kills phone system

Hi Adrian

the issue has only started for me since I changed 'PBX delivers audio' from true to false. Before that everything was working but with bad voice quality and delay.

Was that the same for you?

Is your setup the same/similar as mine?
I'm wondering if between us, we could deduce what the problem is likely to be, by comparing setup?

for instance I'm not sure if its the remote aspect or an rtp port issue or memory issue on the phones, and why do all of the phones dissconnect? - that surely has something to do with 3CX?

I'm considering either a vpn tunnel between my remote 3CX box and the site to make them appear 'local' or buying a server and bring 3CX back on site, but would have liked to get things working as they are.

I am disappointed to say that I signed up for a years 3CX support yesterday only to be told that my setup is non-standard and so in order for them to help, I have to tear up a weeks worth of stressfull and painful config and testing. I'm not asking to lump the responsibility onto them, I just want some advice!
surely they have tried 'hosted' installs where all extensions are external!

this is all extremely frustrating!

thanks for replying

 

[ajlinggi]

Re: attended transfer major problem

All of my grandstreams are boxed up, so cant help you on that front.

But I can tell you that "PBX delivers audio" on my 3cx system has always been set to on. And we still had the issues.

I'd maybe raise a ticket with Grandstream support directly and see what they say.

Voice quality / delay - that sound like your remote extensions dont have enough bandwidth to deal with the call.

Or your bandwidth where you host your 3cx system isnt enough.

if you've got 50 phones all connected externally, you're going to need a pretty hefty upstream at the 3cx end of things.

Cheers,
Adrian.

 

[ajpoole]

Re: attended transfer major problem

Hi Adrian

42Mbps down and 40Mbps on the VPS!

On site we have a dedicated adsl2+ line for VOIP which is 2Mbps up, but we are only using 8 extensions so far so shouldn't be an issue yet, we are waiting for FTTC 40Mbps/10Mbps which should be available before the office fills up with customers.
I'm using G.711 which is 64kps so even with current 2Mbps upstream should still cope with 30 odd simultaneous calls.

We never had the attended tranfer problem whilst PBX delivers audio was true, but it makes sense that you get extra delay, because a ping from 3CX box to on site server is consistently 22ms

Baring in mind, that the broadband and VPS and SIP trunk all come from the same provider and they have an MPLS network, so no calls get routed over the general internet. It should work nicely :S

I have been speaking to a grandstream engineer, who is awaiting wireshark traces from testing on early monday morning.

Thanks

Andrew

 

[farsight]

Re: attended transfer major problem

I fought through issues with external phones on version 8. It was better on version 9. The setup was similar to yours but with fewer phones. I think I ran every variation that I could.

Bottom line, buy a small server and put it on-site. Run as many of the phones as you can using internal IP. Life will be good.

VPN may be an options. I have stayed away from them. It's just another level of encryption -- and delay, that you don't want.

I'm currently running v10 and plan on experimenting with external phones. I use Grandstream phones with no issues.