New to Asterisk: "regected because extension not found"

Asterisk Asterisk Support
1

Hi,

I’m trying to set up a simple SIP inbound system that is managed with adhearsion.
I don’t have any VoIP/softphones/devices that I’m connecting to asterisk, i just want to use an application to interface the incoming call. I don’t particularly even want to dial out. I have looked up documentation for the past 2 days, and I don’t think I’m really understanding the configurations. I have an SIP account that registers fine, but I when I dial into asterisk from my cellphone, the call won’t complete and I get an error.

[color=#FF0000]Error: [Apr 7 19:44:01] NOTICE[20519]: chan_sip.c:19546 handle_request_invite: Call from ‘74275*****’ to extension ‘30355*****’ rejected because extension not found.[/color]

I’ve tried a bunch of different configurations but obviously i’m missing it.
Right now I would just like to call in, and have a soundfile play or something to verify that incoming calls work.
these are my current configurations.

sip.conf

[code]
[general]
dtmfmode = rfc2833
context=general
srvlookup=yes
register => 30355*****:74275*****@sipconnect.ipcomms.net/30355*****
session-timers=refuse
session-expires=180
session-minse=90
session-refresher=uas

[ipcomms]
secret=74275*****
username=30355*****
host=sipconnect.ipcomms.net
allow=ulaw
context=from-pstn
dtmfmode=rfc2833
insecure=invite,port
nat=yes
type=friend
canreinvite=no

; I briefly had this section in user.conf before i started getting this error
; app_macro.c:302 _macro_exec: No such context ‘macro-stdexten’ for macro ‘stdexten’

[30355*****]
secret=74275*****
username=30355*****
host=sipconnect.ipcomms.net
allow=ulaw
context=from-pstn
dtmfmode=rfc2833
insecure=invite,port
nat=yes
type=friend
canreinvite=no[/code]

extensions.conf

[general]
static=yes
writeprotect=no

; i added this after the fact, I realize it's probably completely useless, 
; does asterisk reserve any context names?
[default]
exten => s,1,Verbose(Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()

[from-pstn]
exten => s,1,Wait(1)
exten => s,2,Answer
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,10
exten => s,5,Background(demo-congrats)

Sorry for the nubcake but any help is appreciated.

2

In the context from-pstn, file extensions.conf you should use 30355***** and not s as the destination extension, the last part of your register command in sip.conf, register => 30355*****:74275*****@sipconnect.ipcomms.net/30355***** tells to the system the extension where to send the incoming calls from this peer.
Another way to solve the problem: put s instead of 30355***** at the end of this line .
The context where incoming calls go is is the context parameter defined in the peer section of sip.conf.

Cheers.

Marco Bruni

3

I’ve updated the config and I’m still getting the same error, what am I doing wrong?

sip.conf

[code]
[general]
; I’ve tried putting “context = from-pstn” in here as well
; but that didn’t work
dtmfmode = rfc2833
srvlookup=yes
register => 30355*****:74275*****@sipconnect.ipcomms.net/30355*****
session-timers=refuse
session-expires=180
session-minse=90
session-refresher=uas

[ipcomms]
secret=74275*****
username=30355*****
host=sipconnect.ipcomms.net
allow=ulaw
context=from-pstn
dtmfmode=rfc2833
insecure=invite,port
nat=yes
type=friend
canreinvite=no

[30355*****]
secret=74275*****
username=30355*****
host=sipconnect.ipcomms.net
allow=ulaw
context=from-pstn
dtmfmode=rfc2833
insecure=invite,port
nat=yes
type=peer
canreinvite=no[/code]

extensions.conf

[general]
static=yes
writeprotect=no

[from-pstn]
exten => 30355*****,1,Answer
exten => 30355*****,2,Playback(tt-weasels)
exten => 30355*****,3,Voicemail(44)
exten => 30355*****,4,Hangup
4

After updating extensions.conf did you issue the Asterisk cli command “dialplan reload” ?
Can you post the cli log when a call comes in, to see what happens ?

Cheers.

Marco Bruni

5

Yeah, I used reload and dialplan reload.
With verbosity at 11 the only error I get is

== Using SIP RTP CoS mark 5 [Apr 17 12:18:38] NOTICE[1716]: chan_sip.c:19546 handle_request_invite: Call from '30355*****' to extension '30355*****' rejected because extension not found.