Caller hung up before dial problem

Asterisk Asterisk Dialplan
#1

if i try call, I can see caller hung up before dial
what is the problem?

this is my dial plan
[out1]
exten=>.,1,Set(CALLERID(num)=${cid})
exten=>.,n,Gosub(monitor,${EXTEN},1)
same=>n,Dial(SIP/${EXTEN}@${trunk})
same=>n,Playback(account-balance-is.alaw)

[monitor]
exten=>.,1,set(path=/var/www/html/rec/)
same=>n,Mixmonitor(${path}${STRFTIME(${EPOCH},%Y-%m-%d%H-%M-%S)}-${EXTEN}${type}${src}.wav)
same=>n,return()

[out2]
exten=>.,1,ResetCDR()
exten=>.,n,Set(CALLERID(num)=${cid})
exten=>.,n,Set(CALLERID(name)=${cidname})
exten=>.,n,Playback(account-balance-is.alaw)
same=>n,hangup()

pleaes check and help me
how to do to make dial calling successfull?

#2

You have a channel not implemented error.
Do you have chan_sip.so module loaded?

#3

current chan_sip.so module loaded
and currently in trunk, one pjsip trunk created.
have to create sip trunk instead of pjsip trunk?

#4

If you are using PJSIP, you should keep doing that and not switch to chan_sip.
In the Dial application though, you have to specify PJSIP as the technology instead of SIP.

same => n,Dial(PJSIP/${EXTEN}@${trunk})

https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels

#5

sorry. I did as your guide.
but I can see caller hung up before dial error continually.

#6

when create sip trunk, what information i have to write in peer details?
host=216.53.4.1
username=userid —?
secret=password ----?
type=peer

#7

I found solution
the problem is related with sip trunk.
after created sip trunk correctly, I can connect calling.