Hello
I am trying to establish sipml5 to sipml5 connection using webRTC.
In asterisk user.conf section [1060][1061]
each has these parameters.
type=peer
username=1061
host=dynamic
secret=1061
context=default
hasiax=no
hassip=yes
avpf=yes
encryption=yes
and exten is written like these
exten => 1060,1,Dial(SIP/1060)
exten => 1061,1,Dial(SIP/1061)
Then I tried to call from 1061 to 1060
== Using SIP RTP CoS mark 5
– Executing [1060@default:1] Dial(“SIP/1061-00000000”, “SIP/1060”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/1060
– SIP/1060-00000001 is ringing
Ringing seems to be successful ,1061 plays dial tone and 1060 plays ringtone.
Then push answer button on 1060 and message changes into ‘in call’ ,but 1061 still plays dial tone.
No message appeares in console.
after a few moments,I have to stop ringing by manual operation.
-- No one is available to answer at this time (1:0/0/0)
-- Auto fallthrough, channel 'SIP/1061-00000000' status is 'NOANSWER'
-- Unregistered SIP '1061'
Does anyone success call from sipml5 to sipml5 ?