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Also, get a decent provider, or decent connections from them, we have primary and backup circuits (on different carriers) where the IP is retained on failover.. :-)

I have a Gamma trunk which works absolutely fine with no STUN server" There's your clue right there.... something is rewriting the packets already, turn that off to use STUN

You're in over your head, you should know the answers to your own questions. You don't buy a car then ask some strangers how to drive... P.s This is a terrible setup/proposal :-)

Dial Direct" forces a 2 way call on speakerphone :-)

Easy enough to do with site to site SIP trunks linking the two. Easy that is, if you know how to do them on the IP Office and they know how to on the Cisco. Only way to do it really. Knowing the types of people that do Cisco systems quite well, they aren't very good at the telephony side in my...

It can also burn the ports out and sometimes take a whole card with, pop them and and check them carefully for black soot marks...

No it doesn't, the original source address is passed, it isn't NATd both ways... :-)

Just use IP Routes to only respond to your IP/range, works great :-)

Brown Booby" excellent user name there :-)

That's why he's not called Numpty...:-)

So check your ITSP platform eqipment that provides the trunks just in case, when we've had similar issues it was the provider causing the issue. Also digit duplication is likely not echo, when SIP trunks carry DTMF it is actually removed from the audio stream before transport and then...

It's better to use fallback in ICR, that way you can use the mailbox on the group (while VM is working), you can't do that if it matches a module as it disappears, obviously because the name matches the module :-)

Try to turn on RTP Keep alives on your SYSTEM > LAN > VoIP TAB RTP keepalives: Release 6.1+ These settings can be used with SIP trunks associated with the LAN through their Use Network Topology Info setting (Line | Transport). For some scenarios, with frequent call forwarding on the same...

So assuming you have/want it as follows: Asterisk >>> SIP Trunk on LAN >>> IP Office >>> SIP Trunk on WAN >>> Public Tel Network and not: Asterisk >>> SIP Trunk on WAN >>> Public Tel Network >>> SIP Trunk on WAN >>> IP Office >>> SIP Trunk on WAN >>> Public Tel Network Then all you need on...

I've covered it a few times in the past, search for it, should be there. Obviously calls will be charged to the IP Office if that's where it breaks out :-)

Yes you can, you can take calls in from Asterisk on a trunk and push whatever is dialled to a normal SIP trunk, No attendant or anything needed :-)

You can connect trunk to trunk with SIP on IP Office, no problem. If you mean on the Asterisk, then this is the wrong place to ask :-)

Until you ask them to explain why they reject it you will be nowhere, you have evidence they are. Why don't they just deliver VoIP direct? Seems silly to convert :-)

Avaya do have a tool that coverts them for you, they just don't give it to end users/BPs. I have it, as do a few others, but it's not generally available :-)

Training/knowledge, that's what you're missing [yoda]