Jerry Geis
2011-Apr-04 19:20 UTC
[asterisk-users] 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
Warren Selby
2011-Apr-05 00:26 UTC
[asterisk-users] dialplan is not finding my number asterisk 1.8.3
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110404/eeaa9880/attachment.htm>
Paul Belanger
2011-Apr-05 00:46 UTC
[asterisk-users] dialplan is not finding my number asterisk 1.8.3
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
Mark Deneen
2011-Apr-05 02:06 UTC
[asterisk-users] dialplan is not finding my number asterisk 1.8.3
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? > > JerryI'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
2011-Apr-05 11:44 UTC
[asterisk-users] dialplan is not finding my number asterisk 1.8.3
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