Idea: If something is corrupting your dialplan, then this should reveal the extent of the corruption:
You might, when the system is working properly, do a: asterisk -rx "dialplan show" > somefile1 and then, when you are having problems, do a: asterisk -rx "dialplan show" > somefile2 diff -u somefile1 somefile2 and see if this reveals anything juicy. murf On Tue, Apr 5, 2011 at 5:44 AM, Jerry Geis <[email protected]> wrote: > 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 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Steve Murphy ParseTree Corporation 57 Lane 17 Cody, WY 82414 ✉ [email protected] ☎ 307-899-5535
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
