First, none of this is anything to due with problems with the TrixBox product. It has taken me many days to get my VoIP <> PSTN service back up and going and I will share my story here. All the issues relate to changing many things all at once. Are you sitting comfortably......?

Some  weeks ago my SIP VoIP call quality went really bad; echo, call break up and one-way audio. At the same time I was experiencing internet performance problems so I put it down to that. Later it transpired that a full factory reset of my ADSL router was required. So I reverted back to my Zpatel TDM400P to bridge my Asterisk server to my PSTN line. Since moving to VoIP I had stopped publishing this number as I have always had poor echo issues with the TDM400P and unreliable Calling Line ID (CLID). After many days and many configurations I was unable to fix the echo, in fact it got a lot worse! Now my Wife was very displeased and so I shutdown my TrixBox server and reverted to a DECT regular phone.

Not being one to give up (plus I've paid for UK number and price plan on my VoIP - VoIPTalk.org) I decided to have another go - so hoping that a later Asterisk Build and later Zaptel Build in TrixBox V2 might improve my ZAP issues I wiped the server and started again.

TrixBox installs quickly and easily. I started to set up my IAX and SIP trunks to VoIPTalk and couldn't get any inbound calls working at all. Outbound was OK.

I reconfigured my Grandstream BT102 to connect directly to VoIPTalk via SIP and this worked OK so I knew the service was working OK.

After a day of messing around, and a sneaking suspiscion that nothing was landing on my PBX, I decided to check the ADSL router. Ah Ha! the full factory reset had turned NAT, Security and no ICMP (ping) back on. So even though the router had a fixed public IP address on both sides it was still doing some kind of NAT. Turned these off and suddenly IAX & SIP calls are ariving at my PBX. Now my only problem was that on SIP I could not hear the external caller. I played around with various settings in sip_nat.conf but to no avail. I then started playing with some of the debug options in the Asterisk CLI - sip debug ip to be specific. I noticed that the source IP going back was the loop back address 127.0.0.1 In sip_nat.conf I had added the externhost entry and added the FQDN of my PBX. However, I also have this in /etc/hosts as an alternative to asterisk.local on  127.0.0.1 Once I changed externhost to externip and inserted my fixed public IP then SIP started working properly.

SIP call quality seems good at present. Tomorrow I'll see if ZAP is any better.