Idea:<br><br>If something is corrupting your dialplan, then this should<br>reveal the extent of the corruption:<br><br>You might, when the system is working properly, do a:<br><br>asterisk -rx "dialplan show" > somefile1<br>
<br>and then, when you are having problems, do a:<br><br>asterisk -rx "dialplan show" > somefile2<br>diff -u somefile1 somefile2<br><br>and see if this reveals anything juicy.<br><br>murf<br><br><br><br><div class="gmail_quote">
On Tue, Apr 5, 2011 at 5:44 AM, Jerry Geis <span dir="ltr"><<a href="mailto:geisj@pagestation.com">geisj@pagestation.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Jerry Geis wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
I am calling from a polycom phone into asterisk ( 1105 ) on a PC with a speaker attached.<br>
<br>
When asterisk first starts this works. In fact it works for some time. Then it just stops with this error on the CLI.<br>
<br>
[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'.<br>
<br>
When doing the "dialplan show" it clearly in the context.<br>
<br>
[ Context 'smvoice-mediaport' created by 'pbx_config' ]<br>
'1105' => 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config]<br>
<br>
<br>
Its telling me it cannot find it. Its there - the dialplan shows its there.<br>
When I stop and start it works again for a little while.<br>
Matter of fact I just issued "dialplan reload" and calling into 1105 works again.<br>
<br>
Whats up? How do I get this to be consistent?<br>
<br>
Jerry<br>
<br>
<br>
</blockquote>
I just looked in my extensions.conf and I do not have extenpatternmatchnew at all. My understanding is that<br>
it is off by default.<br>
<br>
my sip.conf has:<br>
register => mndemo_to_mediaport105:secret@mndemo<br>
<br>
; Description:<br>
[mndemo_to_mediaport105]<br>
type=friend<br>
defaultname=mndemo_to_mediaport105<br>
username=mndemo_to_mediaport105<br>
secret=secret<br>
disallow=all<br>
allow=ulaw<br>
allow=alaw<br>
allow=gsm<br>
rtptimeout=60<br>
host=192.168.1.58<br>
context=smvoice-mediaport<br>
<br>
<br>
I was not aware I needed another context of :<br>
<br>
[mndemo_to_mediaport105]<br>
include => smvoice-mediaport<br>
<br>
<br>
The context is set above in sip.conf and that is what the CLI above is showing its using.<br>
<br>
<br>
Also my extensions.conf section is :<br>
<br>
------<br>
[smvoice-mediaport-public-address]<br>
exten => s,1,System(/home/silentm/bin/smfunctions -stop)<br>
exten => s,n,Playback(beep)<br>
exten => s,n,Dial(Console/dsp)<br>
exten => s,n,Hangup<br>
exten => h,1,System(/home/silentm/bin/smfunctions -start)<br>
<br>
[smvoice-mediaport]<br>
exten => public_address,1,Goto(smvoice-mediaport-public-address,s,1)<br>
<br>
#include "/etc/asterisk/express.dnis.conf"<br>
<br>
------<br>
where express.dnis.conf has:<br>
; Phone Caller ID & DNIS Manager screen<br>
<br>
; MMPCGA : VISUAL PC ROOM 105 - exten => 1105,1,Goto(smvoice-mediaport-public-address,s,1)<br>
<br>
-------<br>
Here is a call that works:<br>
== Using SIP RTP CoS mark 5<br>
-- Executing [1105@smvoice-mediaport:1] Goto("SIP/mndemo_to_mediaport105-00000003", "smvoice-mediaport-public-address,s,1") in new stack<br>
-- Goto (smvoice-mediaport-public-address,s,1)<br>
-- Executing [s@smvoice-mediaport-public-address:1] System("SIP/mndemo_to_mediaport105-00000003", "/home/silentm/bin/smfunctions -stop") in new stack<br>
-- Executing [s@smvoice-mediaport-public-address:2] Playback("SIP/mndemo_to_mediaport105-00000003", "beep") in new stack<br>
-- <SIP/mndemo_to_mediaport105-00000003> Playing 'beep.gsm' (language 'en')<br>
-- Executing [s@smvoice-mediaport-public-address:3] Dial("SIP/mndemo_to_mediaport105-00000003", "Console/dsp") in new stack<br>
<< Call placed to 'dsp' on console >> << Auto-answered >> -- Called dsp<br>
-- ALSA/dummy answered SIP/mndemo_to_mediaport105-00000003<br>
-- Executing [h@smvoice-mediaport-public-address:1] System("SIP/mndemo_to_mediaport105-00000003", "/home/silentm/bin/smfunctions -start") in new stack<br>
<< Hangup on console >> == Spawn extension (smvoice-mediaport-public-address, s, 3) exited non-zero on 'SIP/mndemo_to_mediaport105-00000003'<br>
------<br>
<br>
<br>
As I mentioned starting asterisk all this works. There is some random time later - perhaps days where it then stops<br>
finding the exten.<br>
<br>
Is there something I have wrong in the config above?<br>
<br>
Jerry<br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br><div><span style="font-size: 13.3px; font-family: Verdana,Arial,Helvetica,sans-serif;"><div style="margin: 0pt 0pt 8px;"><p style="margin: 0pt;"><span style="font-family: comic sans ms,sans-serif; font-size: medium;">Steve Murphy</span></p>
<p style="margin: 0pt;"><span style="font-family: comic sans ms,sans-serif; font-size: medium;">ParseTree Corporation<br></span></p>
<p style="margin: 0pt;">57 Lane 17</p>
<p style="margin: 0pt;">Cody, WY 82414</p>
<p style="margin: 0pt;">✉ <a href="mailto:murf@parsetree.com" target="_blank">murf@parsetree.com</a></p>
<p style="margin: 0pt;">☎ 307-899-5535</p></div></span><br></div><div><span style="font-size: 13.3px; font-family: Verdana,Arial,Helvetica,sans-serif;"><img src="http://s.wisestamp.com/pixel.png?p=mozilla&v=2.0.3&t=1287885769470&u=6672444&e=5113" width="1" height="1"></span></div>
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