Confbridge() example from definitive guide and Zoiper

I have Asterisk 13.11.2 running and added the confbridge example from page 295 from the definitive guide. I use Zoiper on my iPhone and Macbook and linPhone on my MacBook, just for testing and trying. There seem to be a problem with dtmf which is set to dtmfmode = rfc283 in the [genereal] part of sip.conf and in the soft phones which only occurs when pressing 0 to add another line to the conference. For the rest DTMF is working.

When I press 0 while in the conference I hear a voice prompt telling me to enter the number to call and add to the conference. The digits are received by Asterisk but not used to dial. This is the message that shows up.

-- Executing [conference-joiner@conference:1] NoOp("SIP/5556-00000102", "") in new stack
-- Executing [conference-joiner@conference:2] Read("SIP/5556-00000102", "numberToDial,vm-enter-num-to-call") in new stack
-- <SIP/5556-00000102> Playing 'vm-enter-num-to-call.gsm' (language 'en')
-- User entered '5557'
-- Auto fallthrough, channel 'SIP/5556-00000102' status is 'UNKNOWN'
-- Started music on hold, class 'default', on channel 'SIP/5556-00000102'

5555 is the number of the phone that is used to call the other number.

I’m using autocreatepeers=yes in sip.conf (just a test server without a trunk so there isn’t much that can go wrong.)

Really tried to find what the problem is but I’m running out of ideas what to try and test. Suggestions for correcting mistakes I might make, adjusting the settings of the soft phone, replacement of the soft phones used and/or changes in the Asterisk .conf files are very welcome. I really like this feature and I’m eager to get it up and running in a proper way. Thanks in advance.


; standard participant
exten => 602,1,NoOp()
same => n,Goto(conference,1)

exten => conference,1,NoOp()
    same => n,Set(thisBridge=primary)
    same => n,ConfBridge(${thisBridge},,,volume_ctrl_menu)

exten => conference_joiner,1,NoOp()
    same => n,Read(numberToDial,vm-enter-num-to-call)
    same => n,Originate(SIP/my_itsp/${numberToDial},exten,ConferenceRooms,602,1)

1=increase_listening_volume 4=decrease_listening_volume