asterisk users - Apr 2011 - dialplan is not finding my number asterisk 1.8.3

I am calling from a polycom phone into asterisk ( 1105 ) on a PC with a 
speaker attached.

When asterisk first starts this works. In fact it works for some time. 
Then it just stops with this error on the CLI.

[Apr  4 15:10:21] NOTICE[4357]: chan_sip.c:21358 handle_request_invite: 
Call from 'mndemo_to_mediaport105' to extension '1105' rejected
because
extension not found in context 'smvoice-mediaport'.

When doing the "dialplan show" it clearly in the context.

[ Context 'smvoice-mediaport' created by 'pbx_config' ]
  '1105' =>         1. Goto(smvoice-mediaport-public-address,s,1) 
[pbx_config]


Its telling me it cannot find it. Its there - the dialplan shows its there.
When I stop and start it works again for a little while.
Matter of fact I just issued "dialplan reload" and calling into 1105 
works again.

Whats up? How do I get this to be consistent?

Jerry

On Mon, Apr 4, 2011 at 2:20 PM, Jerry Geis <geisj at pagestation.com>
wrote:
<snip>

> Whats up? How do I get this to be consistent?
>
> Jerry
>
>
Can you post all of the relevant sections of extensions.conf, and the CLI
output of a successful call and the CLI output of a failed called.  The
complete CLI output, from beginning to end of each call.  With this kind of
information we can begin to diagnose what's happening.

-- 
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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On 11-04-04 03:20 PM, Jerry Geis wrote:
> I am calling from a polycom phone into asterisk ( 1105 ) on a PC with a
> speaker attached.
>
> When asterisk first starts this works. In fact it works for some time.
> Then it just stops with this error on the CLI.
>
> [Apr 4 15:10:21] NOTICE[4357]: chan_sip.c:21358 handle_request_invite:
> Call from 'mndemo_to_mediaport105' to extension '1105'
rejected because
> extension not found in context 'smvoice-mediaport'.
>
> When doing the "dialplan show" it clearly in the context.
>
> [ Context 'smvoice-mediaport' created by 'pbx_config' ]
> '1105' => 1. Goto(smvoice-mediaport-public-address,s,1)
[pbx_config]
>
*CLI> dialplan show 1105 at smvoice-mediaport
>
> Its telling me it cannot find it. Its there - the dialplan shows its there.
> When I stop and start it works again for a little while.
> Matter of fact I just issued "dialplan reload" and calling into
1105
> works again.
>
> Whats up? How do I get this to be consistent?
>
Have you included the context properly?

[mndemo_to_mediaport105]
include => smvoice-mediaport


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

On Mon, Apr 4, 2011 at 3:20 PM, Jerry Geis <geisj at pagestation.com>
wrote:
> I am calling from a polycom phone into asterisk ( 1105 ) on a PC with a
> speaker attached.
>
> When asterisk first starts this works. In fact it works for some time. Then
> it just stops with this error on the CLI.
>
> [Apr ?4 15:10:21] NOTICE[4357]: chan_sip.c:21358 handle_request_invite:
Call
> from 'mndemo_to_mediaport105' to extension '1105' rejected
because extension
> not found in context 'smvoice-mediaport'.
>
> When doing the "dialplan show" it clearly in the context.
>
> [ Context 'smvoice-mediaport' created by 'pbx_config' ]
> ?'1105' => ? ? ? ? 1. Goto(smvoice-mediaport-public-address,s,1)
> [pbx_config]
>
>
> Its telling me it cannot find it. Its there - the dialplan shows its there.
> When I stop and start it works again for a little while.
> Matter of fact I just issued "dialplan reload" and calling into
1105 works
> again.
>
> Whats up? How do I get this to be consistent?
>
> Jerry
I'm not all that familiar with 1.8 yet but, with 1.6.2, I ran into
some similar problems with extenpatternmatchnew=yes.  They were
similar in that the dialplan was not executed as expected, but the
behavior was deterministic.  Your experience has things changing over
time which is really quite strange.

-M

Jerry Geis wrote:
> I am calling from a polycom phone into asterisk ( 1105 ) on a PC with 
> a speaker attached.
>
> When asterisk first starts this works. In fact it works for some time. 
> Then it just stops with this error on the CLI.
>
> [Apr  4 15:10:21] NOTICE[4357]: chan_sip.c:21358 
> handle_request_invite: Call from 'mndemo_to_mediaport105' to
extension
> '1105' rejected because extension not found in context 
> 'smvoice-mediaport'.
>
> When doing the "dialplan show" it clearly in the context.
>
> [ Context 'smvoice-mediaport' created by 'pbx_config' ]
>  '1105' =>         1. Goto(smvoice-mediaport-public-address,s,1)
> [pbx_config]
>
>
> Its telling me it cannot find it. Its there - the dialplan shows its 
> there.
> When I stop and start it works again for a little while.
> Matter of fact I just issued "dialplan reload" and calling into
1105
> works again.
>
> Whats up? How do I get this to be consistent?
>
> Jerry
>
>
I just looked in my extensions.conf and I do not have 
extenpatternmatchnew at all. My understanding is that
it is off by default.

my sip.conf has:
register => mndemo_to_mediaport105:secret at mndemo

; Description:
[mndemo_to_mediaport105]
type=friend
defaultname=mndemo_to_mediaport105
username=mndemo_to_mediaport105
secret=secret
disallow=all
allow=ulaw
allow=alaw
allow=gsm
rtptimeout=60
host=192.168.1.58
context=smvoice-mediaport


I was not aware I needed another context of :

[mndemo_to_mediaport105]
include => smvoice-mediaport


The context is set above in sip.conf and that is what the CLI above is showing
its using.


Also my extensions.conf section is :

------
[smvoice-mediaport-public-address]
exten => s,1,System(/home/silentm/bin/smfunctions -stop)
exten => s,n,Playback(beep)
exten => s,n,Dial(Console/dsp)
exten => s,n,Hangup
exten => h,1,System(/home/silentm/bin/smfunctions -start)

[smvoice-mediaport]
exten => public_address,1,Goto(smvoice-mediaport-public-address,s,1)

#include "/etc/asterisk/express.dnis.conf"

------
where express.dnis.conf has:
; Phone Caller ID & DNIS Manager screen

; MMPCGA    : VISUAL PC ROOM 105             - 
exten => 1105,1,Goto(smvoice-mediaport-public-address,s,1)

-------
Here is a call that works:
  == Using SIP RTP CoS mark 5
    -- Executing [1105 at smvoice-mediaport:1]
Goto("SIP/mndemo_to_mediaport105-00000003",
"smvoice-mediaport-public-address,s,1") in new stack
    -- Goto (smvoice-mediaport-public-address,s,1)
    -- Executing [s at smvoice-mediaport-public-address:1]
System("SIP/mndemo_to_mediaport105-00000003",
"/home/silentm/bin/smfunctions -stop") in new stack
    -- Executing [s at smvoice-mediaport-public-address:2]
Playback("SIP/mndemo_to_mediaport105-00000003", "beep") in
new stack
    -- <SIP/mndemo_to_mediaport105-00000003> Playing 'beep.gsm'
(language 'en')
    -- Executing [s at smvoice-mediaport-public-address:3]
Dial("SIP/mndemo_to_mediaport105-00000003", "Console/dsp")
in new stack
 << Call placed to 'dsp' on console >> 
 << Auto-answered >> 
    -- Called dsp
    -- ALSA/dummy answered SIP/mndemo_to_mediaport105-00000003
    -- Executing [h at smvoice-mediaport-public-address:1]
System("SIP/mndemo_to_mediaport105-00000003",
"/home/silentm/bin/smfunctions -start") in new stack
 << Hangup on console >> 
  == Spawn extension (smvoice-mediaport-public-address, s, 3) exited non-zero on
'SIP/mndemo_to_mediaport105-00000003'
------


As I mentioned starting asterisk all this works. There is some random 
time later - perhaps days where it then stops
finding the exten.

Is there something I have wrong in the config above?

Jerry