Hi *
I’ve been struggling for days with this auto-answer-concept, so I really hope someone could put me in the right direction.
I’m still new to asterisk and this is my first attempt to auto-answer a call.
I’ve got a SIP-extension (21) with Digium D60 (with headset) and a SIP-extension (22) softphone (X-Lite).
When I call 21 (from the X-lite), the Digium D60 are just ringing… it’s not automatically answering the call???
Output from asterisk
== Using SIP RTP CoS mark 5
– Executing [21@internal:1] NoOp(“SIP/22-00000083”, “SIP/22-00000083 internal 21”) in new stack
– Executing [21@internal:2] Dial(“SIP/22-00000083”, “SIP/21,b(auto-answer^addheader^1)”) in new stack
== Using SIP RTP CoS mark 5
– SIP/21-00000084 Internal Gosub(auto-answer,addheader,1) start
– Executing [addheader@auto-answer:1] SIPAddHeader(“SIP/21-00000084”, “Alert-Info: ring-answer”) in new stack
[Feb 17 16:36:05] NOTICE[15780][C-00000052]: app_stack.c:1082 gosub_run: SIP/21-00000084 Abnormal ‘Gosub(auto-answer,addheader,1)’ exit. Popping routine return locations.
– Called SIP/21
– SIP/21-00000084 is ringing
My configs:
extension.conf
[general]
extenpatternmatchnew=yes
[globals]
[default]
[auto-answer]
exten => addheader,1,SIPAddHeader(Alert-Info: ring-answer)
[internal]
exten => _2X,1,NoOp(${CHANNEL} ${CONTEXT} ${EXTEN})
same => n,Dial(SIP/${EXTEN},b(auto-answer^addheader^1))
sip.conf
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
allow=gsm
qualify=yes
canreinvite=no
deny=0.0.0.0/0.0.0.0
permit=192.168.0.0/255.255.0.0
[21]
type=friend
secret=mypasswd
callerid=“Digium D60” <21>
host=dynamic
context=internal
[22]
type=friend
secret=mypasswd
callerid=“X-Lite” <22>
host=dynamic
context=internal
Kind regards, Ole