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:
- 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
Regards
Daniel
Hi
You need to post the cli output of the call,
Also I would remove the waits from the dialplan
Ian
Hi
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@192.168.0.105’ 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@192.168.1.179’ 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’
localhost*CLI>
This is with a verbosity of 7.
Regards
Daniel