Call from GSM to Asterisk


#1

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:


#2

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’


#3

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


#4

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