I have a traditional PBX (Alcatel 4100) which I want to integrate with Asterisk. One of the Alcatel analogue extensions (nr 299) is connected to an Atcom AT-100P FXO card (see * below). An incoming rule routes calls coming in on this extension (nr 299) to a SIP phone (Atcom AT-530P) nr 108. From this phone (SIP nr 108) I want the person answering to be able to transfer/forward calls to other extensions (eg 202) on the Alcatel PBX.
I use the following (for testing) in extensions.conf in the default section:
exten => 23,1,Flash()
exten => 23,2,Wait(1)
exten => 23,3,SendDTMF(202,550)
exten => 23,4,Wait(4)
What happens is:
- On the SIP phone (nr 108) the user presses FDW.
- The SIP user then types in 23 and presses send
- The SIP phone (nr. 108) says “Hangup” (i.e. disconnects)
- The original caller gets a busy tone
- After about 3 seconds the Alcatel PABX does put the call through to extension 202 (when that call is answer using the analogue phone the person also has a busy tone).
From this I can deduce that the flash and DTMF does work. My question is how can I prevent the link from being disconnected as explained in step 4 above?
- The AT-100P supports FXS Loopstart and “Kewlstart” (Loopstart with far end disconnection supervision). It can detect ringing and remote hangup and fully supports Pseudo-TDM bridging through Zaptel.
I would appreciate any help on this