Newbie question about Call from '' to extension 'asterisk' r

Asterisk Asterisk Support
1

Hi group,

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.

OBJECTIVE:
I want basic call handling with the system answering, playing a prompt and then hanging up.

SET-UP:
I have the following plan defined in extensions.conf:

[default]
include => incoming

[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:

udpbindaddr=0.0.0.0:5090

[asterisk]
type=incoming
host=dynamic
secret=burp
context=incoming
deny=0.0.0.0/0

When using my SIP phone (Kphone), I get the following output in the terminal (verbose level):

[Feb 20 17:34:08] NOTICE[2687]: 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[2687]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 1212204217@192.168.1.102 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[2687] netsock2.c: == Using SIP RTP CoS mark 5
[Feb 20 17:34:38] NOTICE[2687] chan_sip.c: Call from ‘’ to extension ‘asterisk’ rejected because extension not found in context ‘incoming’.
[Feb 20 17:34:40] WARNING[2687] chan_sip.c: Retransmission timeout reached on transmission 1027795433@192.168.1.102 for seqno 342 (Critical Response)

2

Check the settings of the Softphone, your problem is there.

3

I see nothing wrong with the soft phone.

type=incoming is new to me. Which version of Asterisk is this.

The device entry seems to have matched correctly, and asterisk is correctly reporting that there is no extension called asterisk in the incoming context. You need to add that extension.

4

Had the same problem.

Solution is to use the vmexten setting in your sip extension config.

From the sip.conf sample file

;vmexten=voicemail ; dialplan extension to reach mailbox sets the ; Message-Account in the MWI notify message ; defaults to "asterisk

Your softphone is subscribing for a voicemail notification, and the reply from Asterisk is telling it that the voicemail extension is ‘asterisk’ - the default.

When you use your softphone to access voicemail (sipdroid for me), it dials the provided extension.