I am trying to handle my first incoming SIP call through Asterisk and I have already hit a snag. I looked over the Internet for other poor souls such as myself with similar problems and did not succeed in fixing the problem.
I want basic call handling with the system answering, playing a prompt and then hanging up.
I have the following plan defined in extensions.conf:
include => incoming
exten => s,1,Wait(1)
exten => s,n,Answer()
exten => s,n(goodbye),Playback(vm-goodbye)
exten => s,n(end),Hangup()
In SIP.conf, I have the following:
When using my SIP phone (Kphone), I get the following output in the terminal (verbose level):
[Feb 20 17:34:08] NOTICE: chan_sip.c:21355 handle_request_invite: Call from ‘’ to extension ‘asterisk’ rejected because extension not found in context ‘incoming’.
[Feb 20 17:34:38] WARNING: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission firstname.lastname@example.org for seqno 4259 (Critical Response) – See doc/sip-retransmit.txt.
Packet timed out after 31999ms with no response
And in the full log (not much more information):
[Feb 20 17:34:38] VERBOSE netsock2.c: == Using SIP RTP CoS mark 5
[Feb 20 17:34:38] NOTICE chan_sip.c: Call from ‘’ to extension ‘asterisk’ rejected because extension not found in context ‘incoming’.
[Feb 20 17:34:40] WARNING chan_sip.c: Retransmission timeout reached on transmission email@example.com for seqno 342 (Critical Response)