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@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@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@smvoice-mediaport-public-address:1]
System("SIP/mndemo_to_mediaport105-00000003", "/home/silentm/bin/smfunctions
-stop") in new stack
-- Executing [s@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@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@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
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