Rejected because extension not found in context


#1

I am trying to call from one internal softphone (eyeBeam) to another softphone (Bria - iPhone) with not luck.

The phones are regestering but will not call each other.
eyeBeam is extension 102
Bria is extension 101

Here is the error message that I am getting

[Dec 25 13:46:54] NOTICE[25279]: chan_sip.c:22147 handle_request_invite: Call from ‘102’ (192.168.0.3:1286) to extension ‘101’ rejected because extension not found in context ‘internal’.

here are the sip and extensions .conf file minus the real secret…

sip.conf
[general]
port=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=gsm
callerid=unknown
context=default ;Default context for incoming calls
allowsubscribe=yes
notifyhold=yes
notifyringing=yes
limitonpeer=yes
videosupport=yes
t38pt_udptl=yes ;Default false

[101]
type=friend
context=internal
host=dynamic
secret=1234
mailbox=100@default

[102]
type=friend
context=internal
host=dynamic
secret=1234

extension.conf

[general]
static=yes
writeprotect=yes
priorityjumping=no
autofallthrough=no

[globals]
OFFICE_OPEN_OVERRIDE=
#include trunks.include
DIALOUT=9
INTERNATIONAL-PREFIX=011
RINGTIME=30
TL_DASH=-
TL_MULTI=1
OPERATOR=0
RECORDING_FORMAT=WAV
TL_ENABLE_MAXCALLS_CHECK=0
autofallthrough=yes

[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()

[incoming_calls]

[internal]
exten => s,1,Verbose(1|Echo test application)
exten => s,n,Echo()
exten => s,n,Hangup()

exten => 101,1,Log(NOTICE,“101 ACCOUNT”)
exten => 101,n,Dial(SIP/101,120,Tt)

exten => 102,1,Log(NOTICE,“102 ACCOUNT”)
exten => 102,n,D,Dial(SIP/102,120,Tt)

[phones]
include => internal
include => default

after I do modifications to the files I reload and do a sip reload.

CLI> sip reload
Reloading SIP
== Parsing ‘/etc/asterisk/sip.conf’: == Found
== Parsing ‘/etc/asterisk/users.conf’: == Found
== Using SIP CoS mark 4
== Parsing ‘/etc/asterisk/sip_notify.conf’: == Found

I notice that it reload te users.config. what are the modifications I need to do on that?

Ubuntu server 11.10
Asterisk 1.8x


#2

you need a dialplan reload to take effect your modifications on extensions.conf


#3

did a dialplan reload

I get this at the CLI

== Using SIP RTP CoS mark 5
– Executing [102@internal:1] Log(“SIP/101-00000003”, “NOTICE,“102 ACCOUNT””) in new stack
[Dec 26 11:25:20] NOTICE[20664]: Ext. 102:1 @ internal: “102 ACCOUNT”
[Dec 26 11:25:20] WARNING[20664]: pbx.c:4088 pbx_extension_helper: No application ‘D,Dial’ for extension (internal, 102, 2)
== Spawn extension (internal, 102, 2) exited non-zero on ‘SIP/101-00000003’

The Bria phone shows an error of decline (603)


#4

hi,
exten => 102,n,D,Dial(SIP/102,120,Tt)
should be
exten => 102,n,Dial(SIP/102,120,Tt)


#5

That seems to have worked!!!

Now just need to figure out why there is now audio…