Fail in incoming call

Asterisk Asterisk Dialplan
1

Hi guys! Can you help me. I have a web interface connected with asterisk by websockets! Outgoing calls works perfectly!
I have a 1 external and 1 internal account
each account has own dialplan
Incoming dialplan is:
[1018] //internal

[10018] //external

exten => _XXXXXXXXXX,1,Dial(SIP/1018)

== Using SIP RTP CoS mark 5
– Executing [XXXXXXXXXX@crm_in18:1] Dial(“SIP/1000018-0000000e”, “SIP/1000018”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/1000018
– SIP/1000018-0000000f redirecting info has changed, passing it to SIP/1000018-0000000e
– SIP/1000018-0000000f is busy
== Everyone is busy/congested at this time (1:1/0/0)
– Auto fallthrough, channel ‘SIP/1000018-0000000e’ status is ‘BUSY’

I don’t know why it’s busy! Can you advice me something? Thanks!

2

The call is going into context “crm_in18” and resulting in the dialing of SIP/1000018 which is the same as the calling channel. It responds with busy presumably because it is in a call.

3

Yes you are right! I’m miss one zero and sending it to a same channel as called!

I change it and now it writes
== Using SIP RTP CoS mark 5
– Executing [9661742032@crm_in18:1] Dial(“SIP/1000018-0000001c”, “SIP/1018”) in new stack
– No devices or endpoints to dial (technology/resource)
– Auto fallthrough, channel ‘SIP/1000018-0000001c’ status is ‘CHANUNAVAIL’

Why this?

4

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