Unable to create channel of type 'SIP'(cause 20 - Subscribe)

Asterisk Asterisk Support
1

Hi!

I’m triying to make call from my telephone(SIP/3020) to my softphone(SIP/3028) using ATA Cisco SPA 8000 is ok, the call is made.

But the problem is, if I make call from my softphone(SIP/3028) to my telephone(SIP/3020) is not possible.

Please, see by CLI

== Using SIP RTP CoS mark 5
    -- Executing [3028@ramais:1] Dial("SIP/3020-00000013", "SIP/3028,60,tT") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/3028
    -- SIP/3028-00000014 is ringing
    -- Got SIP response 486 "Busy Here" back from XXX.X.XXX.X:36980
    -- SIP/3028-00000014 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Auto fallthrough, channel 'SIP/3020-00000013' status is 'BUSY'
  == Using SIP RTP CoS mark 5
    -- Executing [3020@ramais:1] Dial("SIP/3028-00000015", "SIP/3020,60,tT") in new stack
[Apr 30 16:39:10] WARNING[3972][C-0000000f]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/3028-00000015' status is 'CHANUNAVAIL'

My Asterisk server is in one network, and my ATA and softphone in other.

Is a problem in my NAT configuration?

Thanks in advanced!

2

The phone has not registered.