Call from GSM to Asterisk

Hi everyone!
I cannot reach my SIP phone from my mobile phone. I´m using 2N VoiceBlue GSM Gateway.
When I call from SIP phone to mobile phone it works correctly, but in other way (GSM -> SIP phone) it cannot reach and I´m getting crazy about that cause I didn´t know why.
everytime asterisk send me this answer:

[Dec 20 18:37:32] NOTICE[2566][C-00000017]: chan_sip.c:26455 handle_request_invite: Call from 'SIP_Trunk_CUCM_210' (10.22.19.210:5060) to extension '102' rejected because extension not found in context 'default'.

But I don´t use any default context. It´s never ever exist

here is my setting:
sip.conf

[general]
bindport=5060
srvlookup=yes
qualify=1000
dtmfmode=rfc2833
language=cs
allowguest=yes


[SIP_Trunk_CUCM_210]
type=peer
cotnext=trunk

host=10.22.19.210
allow=ulaw
allow=gsm

[102]
type=friend
context=trunk
;context=default

host=dynamic
username=102
secret=102
callerid=SWtel_Adam

extensions.conf

[trunk]
;SIP_Trunk_CUCM_210
exten => _6XXXXXXXX,1,Dial(SIP/${EXTEN:0}@10.22.19.210,,r)

exten => _7XXXXXXXX,1,Dial(SIP/${EXTEN:0}@10.22.19.210,,r)

exten => 102,1,Dial(SIP/102)

In wireshark I see INVITE from sip:00420601392648@10.22.19.210 to sip:102@10.48.67.66 on my asterisk server, so there is correct connection.

Help me please :slight_smile:

If you have guest turned on and the request does not match a defined peer then the call will land in the guest context which is by default ‘default’

Issue here is you have context name mistyped and for tha reason Asterisk uses the default context as the context value for the peer SIP_Trunk_CUCM_210

Thanks a lot man! I was blind like a lemon :smiley: