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

it is my first dial plan hello world
i made it from the manual book of Asterisk
i hear hello world and everything is ok but i saw this message appear

[Jul 29 01:35:55] WARNING[5769][C-00000007]: app_dial.c:2437 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)

when i make sip show peers

dhcppc3*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
5001 (Unspecified) D Yes Yes 0 Unmonitored
5002 (Unspecified) D Yes Yes 0 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline]

i notice that host is (Unspecified)
i tried so many times to make it read ip but no way

BUT IT IS RING AND MAKE MESSAGE HELLO WORLD

so , is this a problem or i let it as it is ???

enable the qualify on your sip peers, and run sip show peers again, because the messaged above indicate the peer is not registered

i did and reload after editing sip.conf as shownbut it still as it is …

the sip file is …

[general]
context=unauthenticated ; default context for incoming calls
allowguest=yes ; disable unauthenticated calls
srvlookup=no ; disable DNS SRV record lookup on outbound calls
qualify=yes
; (unless you have a reliable DNS connection,
; in which case yes)
udpbindaddr=0.0.0.0 ; listen for UDP requests on all interfaces
tcpenable=no ; disable TCP support
office-phone ; create a template for our devices
type=friend
; IP second
context=LocalSets ; this is where calls from the device will enter
; the dialplan
host=dynamic ; the device will register with asterisk
nat=force_rport,comedia ; assume device is behind NAT
; *** NAT stands for Network Address Translation,
; which allows multiple internal devices to share an
; external IP address.
dtmfmode=auto ; accept touch-tones from the devices, negotiated
; automatically
disallow=all ; reset which voice codecs this device will accept or offer
allow=g722 ; audio codecs to accept from, and request to, the device
allow=ulaw ; in the order we prefer
allow=alaw
; define a device name and use the office-phone template
5001
secret=5001 ; a unique password for this device –
; DON’T USE THE PASSWORD WE’VE USED IN THIS EXAMPLE!
; define another device name using the same template
5002
secret=5002 ; a unique password for this device –
; DON’T USE THE PASSWORD WE’VE USED IN THIS EXAMPLE!i


dhcppc3CLI> module reload
[Jul 29 02:40:27] NOTICE[6849]: app_queue.c:7824 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action.
[Jul 29 02:40:27] NOTICE[6849]: cel_custom.c:95 load_config: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Jul 29 02:40:27] WARNING[6849]: pbx.c:9902 add_priority: Unable to register extension ‘5002’ priority 1 in ‘LocalSets’, already in use
dhcppc3
CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
5001 (Unspecified) D Yes Yes 0 UNKNOWN
5002 (Unspecified) D Yes Yes 0 UNKNOWN
2 sip peers [Monitored: 0 online, 2 offline Unmonitored: 0 online, 0 offline]
dhcppc3*CLI>

There is no address information of how to reach the devices. They haven’t registered. You’d need to use “sip set debug on” to see if their REGISTER is reaching Asterisk, if not then the problem is outside of Asterisk.

Try to reboot the phone. I ran into a strange situation when i can make call from one phone to another but getting this error when trying to make call in reverse direction.