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

Asterisk Asterisk Support
1

I configured the asterisk and two users 8001 and 8002. When I call from the 8001 to 8002 the it works well but when i call from 8002 to 8001 then call not send to 8001.

8001 and 8002 both are registered successfully when i made the call.

sip.conf

[8001]
type=friend
host=dynamic
secret=8001
context=users

[8002]
type=friend
host=dynamic
secret=8002
context=users

== Parsing ‘/etc/asterisk/sip.conf’: Found
== Parsing ‘/etc/asterisk/users.conf’: Found
== Using SIP CoS mark 4
== Parsing ‘/etc/asterisk/sip_notify.conf’: Found
== Using SIP RTP CoS mark 5
– Executing [8001@users:1] Answer(“SIP/8002-0000001a”, “”) in new stack
> 0x7ff09c046970 – Probation passed - setting RTP source address to 192.168.1.129:8000
– Executing [8001@users:2] Dial(“SIP/8002-0000001a”, “SIP/8001,60”) in new stack
[Dec 12 17:34:42] WARNING[8402][C-00000012]: 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)
– Executing [8001@users:3] VoiceMail(“SIP/8002-0000001a”, “8001@vm”) in new stack
– <SIP/8002-0000001a> Playing ‘vm-intro.gsm’ (language ‘en’)
– <SIP/8002-0000001a> Playing ‘beep.gsm’ (language ‘en’)
– Recording the message
– x=0, open writing: /var/spool/asterisk/voicemail/vm/8001/tmp/WFQBhf format: wav49, 0x7ff0a0005a88
– x=1, open writing: /var/spool/asterisk/voicemail/vm/8001/tmp/WFQBhf format: gsm, 0x7ff0a001f6b8
– x=2, open writing: /var/spool/asterisk/voicemail/vm/8001/tmp/WFQBhf format: wav, 0x7ff0a001fcd8
– User hung up
== Spawn extension (users, 8001, 3) exited non-zero on 'SIP/8002-0000001a’
server2*CLI>

2

Device 8001 hasn’t registered.

It can still make calls because you have used type=friend, rather than type=peer. You might also have allowguest enabled.

3

I double check 8001 device is already registered.

4

The error message is saying otherwise.