Re: [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@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_mediaport105-0003, 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_mediaport105-0003, /home/silentm/bin/smfunctions -stop) in new stack -- Executing [s@smvoice-mediaport-public-address:2] Playback(SIP/mndemo_to_mediaport105-0003, beep) in new stack -- SIP/mndemo_to_mediaport105-0003 Playing 'beep.gsm' (language 'en') -- Executing [s@smvoice-mediaport-public-address:3] Dial(SIP/mndemo_to_mediaport105-0003, Console/dsp) in new stack Call placed to 'dsp' on console Auto-answered -- Called dsp -- ALSA/dummy answered SIP/mndemo_to_mediaport105-0003 -- Executing [h@smvoice-mediaport-public-address:1] System(SIP/mndemo_to_mediaport105-0003, /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_mediaport105-0003' -- 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
Re: [asterisk-users] dialplan is not finding my number asterisk 1.8.3
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 ge...@pagestation.com 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_mediaport105-0003, 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_mediaport105-0003, /home/silentm/bin/smfunctions -stop) in new stack -- Executing [s@smvoice-mediaport-public-address:2] Playback(SIP/mndemo_to_mediaport105-0003, beep) in new stack -- SIP/mndemo_to_mediaport105-0003 Playing 'beep.gsm' (language 'en') -- Executing [s@smvoice-mediaport-public-address:3] Dial(SIP/mndemo_to_mediaport105-0003, Console/dsp) in new stack Call placed to 'dsp' on console Auto-answered -- Called dsp -- ALSA/dummy answered SIP/mndemo_to_mediaport105-0003 -- Executing [h@smvoice-mediaport-public-address:1] System(SIP/mndemo_to_mediaport105-0003, /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_mediaport105-0003' -- 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 ✉ m...@parsetree.com ☎ 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
Re: [asterisk-users] dialplan is not finding my number asterisk 1.8.3
Steve Murphy wrote: 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 Steve, That is a great idea. I did that the first time it happened. I dumped the dialplan, then I restarted and dumped again. it was the same. Being the first time I thought it was just a fluke but now it has happened a couple of times. I have not been able to narrow anything down. Thanks, 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
Re: [asterisk-users] dialplan is not finding my number asterisk 1.8.3
Jerry Geis wrote: Steve Murphy wrote: 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 Steve, That is a great idea. I did that the first time it happened. I dumped the dialplan, then I restarted and dumped again. it was the same. Being the first time I thought it was just a fluke but now it has happened a couple of times. I have not been able to narrow anything down. Thanks, jerry Steve, perhaps I did something wrong the first time. As I just got the error again. I dumped the dialplan and my section: [ Context 'smvoice-mediaport' created by 'pbx_config' ] is empty. when I restart and dump again. [ Context 'smvoice-mediaport' created by 'pbx_config' ] '1105' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] 'mediaport_direct' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] 'public_address' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] I have the correct data. The only thing I have in the dialplan for this box 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) Can a system call be removing stuff from the dialplan? What next? 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
Re: [asterisk-users] dialplan is not finding my number asterisk 1.8.3
Oh, you are *not* going to like this, but you have a few different paths: 1. If the dialplan stuff is not really a memory corruption, but some sort of unplanned, but maybe accidentally programmed thing, either by you or something in the asterisk code, then: a. compile asterisk for debug. You can get in the menuselect stuff and make sure the debug compile options are turned on. Install it. b. Shut down asterisk c. fire it back up, under gdb: gdb full path to asterisk br ast_context_remove_extension_callerid2 comm 1 where c end run normal arguments to asterisk Then use asterisk as normal and wait for the problem to re-occur. Look to see if any calls to ast_context_remove_extension_callerid2 occurred (they will occur with dial reloads-- lots of them). You'll have to search carefully! The gdb messages could be buried in your console output. If the problem reoccurs, but you didn't see any calls to ast_context_remove_extension_callerid2, then it could be assumed that you are suffering some sort of memory corruption. Where it dies, or things start looking strange, may not indicate where or why the corruption is happening. In such a case, it really gets tricky to debug. Then we might resort to stuff like dmalloc, and others, to help spot where/when corruption occurs. Let's cross that bridge if we come to it. murf On Tue, Apr 5, 2011 at 7:30 AM, Jerry Geis ge...@pagestation.com wrote: Jerry Geis wrote: Steve Murphy wrote: 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 Steve, That is a great idea. I did that the first time it happened. I dumped the dialplan, then I restarted and dumped again. it was the same. Being the first time I thought it was just a fluke but now it has happened a couple of times. I have not been able to narrow anything down. Thanks, jerry Steve, perhaps I did something wrong the first time. As I just got the error again. I dumped the dialplan and my section: [ Context 'smvoice-mediaport' created by 'pbx_config' ] is empty. when I restart and dump again. [ Context 'smvoice-mediaport' created by 'pbx_config' ] '1105' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] 'mediaport_direct' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] 'public_address' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] I have the correct data. The only thing I have in the dialplan for this box 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) Can a system call be removing stuff from the dialplan? What next? Oh, you are *not* going to like this, but you have a few different paths: 1. If the dialplan stuff is not really a memory corruption, but some sort of unplanned, but maybe accidentally programmed thing, either by you or something in the asterisk code, then: a. compile asterisk for debug. You can get in the menuselect stuff and make sure the debug compile options are turned on. Install it. b. Shut down asterisk c. fire it back up, under gdb: gdb full path to asterisk br ast_context_remove_extension_callerid2 comm 1 where c end run normal arguments to asterisk Then use asterisk as normal and wait for the problem to re-occur. Look to see if any calls to ast_context_remove_extension_callerid2 occurred (they will occur with dial reloads-- lots of them). You'll have to search carefully! The gdb messages could be buried in your console output. If the problem reoccurs, but you didn't see any calls to ast_context_remove_extension_callerid2, then it could be assumed that you are suffering some sort of memory corruption. Where it dies, or things start looking strange, may not indicate where or why the corruption is happening. In such a case, it really gets tricky to debug. Then we might resort to stuff like dmalloc, and others, to help spot where/when corruption occurs. Let's cross that bridge if we come to it. murf Jerry -- Steve Murphy ParseTree Corporation -- _ -- 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
Re: [asterisk-users] dialplan is not finding my number asterisk 1.8.3
On Mon, Apr 4, 2011 at 2:20 PM, Jerry Geis ge...@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 -- _ -- 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
Re: [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@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 -- _ -- 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
Re: [asterisk-users] dialplan is not finding my number asterisk 1.8.3
On Mon, Apr 4, 2011 at 3:20 PM, Jerry Geis ge...@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 -- _ -- 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
