Asterisk cuts connection before call transfer takes place

Hi there

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:

  1. On the SIP phone (nr 108) the user presses FDW.
  2. The SIP user then types in 23 and presses send
  3. The SIP phone (nr. 108) says “Hangup” (i.e. disconnects)
  4. The original caller gets a busy tone
  5. 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




You need to post the cli output of the call,
Also I would remove the waits from the dialplan



Thanks for the reply. I have removed the waits but it still does not work.

The CLI output is as follows.

Really destroying SIP dialog ‘76e99eb13ca6b61b2c9730d27c2c7e10@’ Method: NOTIFY
– Starting simple switch on ‘Zap/1-1’
[Sep 27 19:18:38] NOTICE[22695]: chan_zap.c:6364 ss_thread: Got event 18 (Ring Begin)…
[Sep 27 19:18:38] NOTICE[22695]: chan_zap.c:6364 ss_thread: Got event 2 (Ring/Answered)…
[Sep 27 19:18:40] NOTICE[22695]: chan_zap.c:6364 ss_thread: Got event 18 (Ring Begin)…
– Executing [s@DID_trunk_1:1] Goto(“Zap/1-1”, “default|108|1”) in new stack
– Goto (default,108,1)
– Executing [108@default:1] Macro(“Zap/1-1”, “stdexten|108|SIP/108”) in new stack
– Executing [s@macro-stdexten:1] Dial(“Zap/1-1”, “SIP/108|20”) in new stack
– Called 108
– SIP/108-08625a00 is ringing
– SIP/108-08625a00 answered Zap/1-1
– Started music on hold, class ‘default’, on Zap/1-1
– Stopped music on hold on Zap/1-1
== Spawn extension (numberplan-custom-2, 42, 0) exited non-zero on ‘Zap/1-1’
– Executing [42@numberplan-custom-2:1] Flash(“Zap/1-1”, “()”) in new stack
Really destroying SIP dialog ‘2144430753-2983417694@’ Method: REGISTER
– Flashed channel Zap/1-1
– Executing [42@numberplan-custom-2:2] SendDTMF(“Zap/1-1”, “201”) in new stack
== Auto fallthrough, channel ‘Zap/1-1’ status is ‘ANSWER’
– Hungup 'Zap/1-1’

This is with a verbosity of 7.