asterisk users - Jan 2011 - Asterisk extension not found problem...

Hi All,

I am using Asterisk for one of my projects in OpenBTS. I am having the age
old problem of "extension not found" when try to make
a call from one registered SIP phone to other registered SIP phone (two
mobile phones connected to Asterisk via OpenBTS).

The exact error thrown on Asterisk CLI is
*"chan_sip.c:20039 handle_request_invite: Call from [IMSI310410270465840]
to
extension "2103" rejected because extension not found"*

I have provisioned for both the phones in *sip.conf* and
*extensions.conf*under context
* [sip-external]* but I suspect whatever entry given in extensions.conf,
that file is not getting parsed and extensions are not read.

I have tried all the methods suggested by others in the Asterisk User
community but still the problem remains same. If anybody knows the solution
to this
one, please let me know.

--
Abhinav


Copied below is my sip.conf and extensions.conf
==================================
*extensions.conf*
==============================[globals]

;Using this Macro
[macro-dialGSM]
exten => s,1,Dial(SIP/${ARG1})
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-CANCEL,1,Hangup
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Busy(30)
exten => s-CONGESTION,1,Congestion(30)
exten => s-CHANUNAVAIL,1,playback(ss-noservice)
exten => s-CANCEL,1,Hangup

#include "extensions.local.conf"

[sip-external]
exten => 2101,1,Macro(dialGSM,2101)
exten => 2102,1,Macro(dialGSM,IMSI310410270465840)
exten => 2103,1,Macro(dialGSM,IMSI404864430002302)

; check for local extensions first
include => sip-local
==============================
*sip.conf*
=============================[general]
; Comment these out if no backhaul is available.
; Use the pair with the shortest latency.
;register => kestrel0:v01ptest at sip.ca1.link2voip.com:5060
;register => kestrel0:v01ptest at sip.ca2.link2voip.com:5060
;register => kestrel0:v01ptest at sip.us1.link2voip.com:5060
;register => kestrel0:v01ptest at sip.us2.link2voip.com:5060
;register => kestrel0:v01ptest at sip.nl1.link2voip.com:5060
;register => kestrel0:v01ptest at sip.nl2.link2voip.com:5060
rtpstart=16386
rtpend=16482
relaxdtmf=yes


[softPhone]
callerid=2101
canreinvite=no
type=friend
context=sip-external
allow=ulaw
allow=gsm
host=dynamic

; provisioned Thu Dec 13 17:15:10 2010
[IMSI310410270465840] ; ATnT SIM card IMSI
callerid=2102
canreinvite=no
type=friend
context=sip-external
allow=gsm
host=dynamic
dtmfmode=info

; provisioned Thu Dec 14 12:15:10 2010
[IMSI404864430002302] ; Vodafone SIM card IMSI
callerid=2103
canreinvite=no
type=friend
context=sip-external
allow=gsm
host=dynamic
dtmfmode=info
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On 11-01-18 08:52 PM, abhinav anand wrote:
> I have tried all the methods suggested by others in the Asterisk User
> community but still the problem remains same. If anybody knows the solution
> to this
> one, please let me know.
> 
Which context is your incoming calls using? When you know that, you can run:

*CLI> dialplan show 2103@<incoming context>

to see if the dialplan actual exists.

-- 
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org

On Tue, 18 Jan 2011, abhinav anand wrote:
> The exact error thrown on Asterisk CLI is "chan_sip.c:20039 
> handle_request_invite: Call from [IMSI310410270465840] to extension 
> "2103" rejected because extension not found"
What context does 'sip show user IMSI310410270465840' show?

What does 'dialplan show 2103@<context-from-previous-command>'
show?

-- 
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867 PST
Newline                                              Fax: +1-760-731-3000