[Asterisk-Users] PBX-VPN-SIP-Asterisk trouble

[Martin Joseph]

On Mar 7, 2006, at 7:02 AM, artifex maximus wrote:

> Hi all!
>
> I have the following setup:
> Phone lines -> traditional PBX -> Welltech 3802
> -> VPN ->
> Asterisk -> Linksys PAP2/Welltech ATA-151 -> phone
>
> There is 2 pieces of Welltech 3802 (2 port FXO) connected to 4 (2x2)
> PBX extensions. Asterisk is a proxy here. Each device successfully
> register itself. I tried the setup above with Linksys and Welltech
> devices as well.
>
> I setup Asterisk as a local PBX and phones can call each others on
> Asterisk side and possible transfering calls. I setup Welltech 3802
> with hotline mode so if someone call the public number from outside
> the call transferred through VPN and phone rings in front of me.
> Great. It's still possible transfer call within Asterisk side.
> Excellent.
>
> The problem comes when I want to call extension on PBX side or
> transfer incoming call to the PBX side. I got the line sound when I
> press flash, the caller hear the MOH and when I call extensions on PBX
> side I got only busy tone.
If I am not mistaken, then I think your question is regarding the flash 
event on the welltech hardware and getting that to work correctly for 
switching in a "call waiting" situation?  Although your setup is MUCH 
more complex then mine,  I have this problem with the welltech 3701a as 
well.
>
> How could I tell that Asterisk send back the flash&DTMF on the same
> PBX extension where call comes from? I think this is important for PBX
> to connect lines "inside" right. How could I route outcoming calls on
> a port of Welltech 3802?
The outgoing calls on the welltech (at least for the SIP setup) are 
routed via the welltechs setup screens which include a totally cryptic 
"routing setup" or some such.  Try searching this list for the last 
couple of days for "welltech" I think someone explained this nicely.
> <snip>
Sorry I am not helpful on this, but looking for a similar answer?