Re: [asterisk-users] dialplan goto - bad priority
On 9/6/2021 9:22 AM, marek wrote:
> hi,
>
> i have this dialplan
>
> [incoming]
> exten => _X./_+421X,1,noop(cut +421 from CALLER)
> exten => _X./_+421X,n,Set(CALLERID(num)=${CALLERID(num):4})
> exten => _X./_+421X,n,goto(${CONTEXT},${EXTEN},1)
> exten => _X.,1,noop(main block)
> exten => _X.,n,noop(main block #2)
> exten => _X.,n,Set(GWNAME=out)
>
> my problem is when call arrive with +421 so i want strip this example
> prefix from callerid
>
> then i expecting that call jump(goto) to the line
>
> exten => _X.,1,noop(main block)
>
> but it jumps to
>
> exten => _X.,n,Set(GWNAME=out)
>
> any idea of this behavior?
That is the intended behavior. See
https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching#PatternMatching-MatchingonCallerID.
When you change the Caller ID and you are extension pattern matching on
that, the extension immediately changes. Thus, you will advance to the
next priority, but in a different extension. You never hit
goto(${CONTEXT},${EXTEN},1) because as soon as you change the Caller
ID, it's not going to pattern match on that priority anymore. It's going
to fall back to the more general extension match, with the next priority.
A simple way to avoid this might be rewriting using a separate context:
[rewrite-421]
exten => _X!,1,NoOp(cut +421 from CALLER)
same => n,Set(CALLERID(num)=${CALLERID(num):4})
same => n,Goto(incoming,${EXTEN},1)
[incoming]
exten => _X./_+421X,1,Goto(rewrite-421,${EXTEN},1)
exten => _X.,1,noop(main block)
exten => _X.,n,noop(main block #2)
exten => _X.,n,Set(GWNAME=out)
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[asterisk-users] dialplan goto - bad priority
hi,
i have this dialplan
[incoming]
exten => _X./_+421X,1,noop(cut +421 from CALLER)
exten => _X./_+421X,n,Set(CALLERID(num)=${CALLERID(num):4})
exten => _X./_+421X,n,goto(${CONTEXT},${EXTEN},1)
exten => _X.,1,noop(main block)
exten => _X.,n,noop(main block #2)
exten => _X.,n,Set(GWNAME=out)
my problem is when call arrive with +421 so i want strip this example
prefix from callerid
then i expecting that call jump(goto) to the line
exten => _X.,1,noop(main block)
but it jumps to
exten => _X.,n,Set(GWNAME=out)
any idea of this behavior?
Marek
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Re: [asterisk-users] Dialplan - using multiple AND or OR in set is it possible ?
Use the ARRAY version of Set. same = n,ExecIf($["A" = "B"]?Set(ARRAY(C,D)=1,2)) On Tue, Apr 21, 2020 at 3:56 AM Administrator wrote: > Hello, > > we want to use something like > > same = n,ExecIf($["A" = "B"]?Set(C=1) & Set(D=2) & ...) > > Problem is that result gives C=1) & Set(D=2) & ... > > Is there a possibility to use multiple AND or OR in such a way ? > > -- > Daniel > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan - using multiple AND or OR in set is it possible ?
Le 21/04/2020 à 15:23, Antony Stone a écrit : On Tuesday 21 April 2020 at 12:54:49, Administrator wrote: Hello, we want to use something like same = n,ExecIf($["A" = "B"]?Set(C=1) & Set(D=2) & ...) Problem is that result gives C=1) & Set(D=2) & ... Is there a possibility to use multiple AND or OR in such a way ? No, logical operators are for comparing True and False - they can't be used to say "do multiple things". I'd suggest two ways of doing what you need: a) invert the test and change the ExecIf() to a GotoIf() which skips past the next few lines, each of which has one of your Set() statements on it. b) leave the logic as it is but change ExecIf() to GosubIf) and put the Set() statements into a subroutine context. Thanks for your reply. We had applied the second approach. Regards -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan - using multiple AND or OR in set is it possible ?
On Tuesday 21 April 2020 at 12:54:49, Administrator wrote: > Hello, > > we want to use something like > > same = n,ExecIf($["A" = "B"]?Set(C=1) & Set(D=2) & ...) > > Problem is that result gives C=1) & Set(D=2) & ... > > Is there a possibility to use multiple AND or OR in such a way ? No, logical operators are for comparing True and False - they can't be used to say "do multiple things". I'd suggest two ways of doing what you need: a) invert the test and change the ExecIf() to a GotoIf() which skips past the next few lines, each of which has one of your Set() statements on it. b) leave the logic as it is but change ExecIf() to GosubIf) and put the Set() statements into a subroutine context. Regards, Antony. -- René Descartes walks in to a bar. The barman asks him "Do you want a drink?" Descartes says "I think not," and disappears. Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialplan - using multiple AND or OR in set is it possible ?
Hello, we want to use something like same = n,ExecIf($["A" = "B"]?Set(C=1) & Set(D=2) & ...) Problem is that result gives C=1) & Set(D=2) & ... Is there a possibility to use multiple AND or OR in such a way ? -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan reload from AMI
Does reloading pbx_config ONLY reload the dialplan? Or is something else reloaded too? This sounds like a preferable way to do it From: Ian McMaster [mailto:ian.mcmas...@gmail.com] Sent: Saturday, April 20, 2019 1:19 PM Subject: Dialplan reload from AMI Rather than Action: Command Command: dialplan reload Prefer this: Action: Reload Module: pbx_config -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan reload not showing debug info even with debug on (ast 15.5)
On Sun, Jul 29, 2018 at 10:04 AM, Jonathan H wrote: > OK, many thanks for that. Not sure I see the point of the change, but > at least I can get the info back by changing > > console => notice,warning,error > to > console => notice,warning,error,debug > > That said, dialplan reload seems to show significantly fewer items > than before. I've got loads of extensions, and it does indeed seem to > reload them all, but only shows about 1/4 of them in the console when > doing dialplan reload. Hmmm... > > Incidentally, just while I'm here, is there a particular reason that > debug can only every be pushed higher when connecting to the console? > it's always been the case since I started using Asterisk 3 years ago > so I guess there's a reason, and I never questioned it before. I'm > just curious! > For example: > > > asterisk -rvddd > Core debug was 2 and is now 3. > > asterisk -rv > Core debug was 3 and is now 4. > > asterisk -rvdd > Core debug is still 4. > > asterisk -rvd > Core debug is still 4. > > But it always respects "core set debug" in whichever direction of > verbosity is required. > When you connect to a remote asterisk with the -r option, there are a couple commands automatically sent every time. These automatic commands tell the remote asterisk what verbose and debug level you passed on the command line. core set verbose at least X silent core set debug atleast X That is why the debug level does not go down. Another thing is that the debug level is global to the system. Thus if you set the level in one connection it affects all connections including future ones. The verbose level is per connection. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan reload not showing debug info even with debug on (ast 15.5)
OK, many thanks for that. Not sure I see the point of the change, but at least I can get the info back by changing console => notice,warning,error to console => notice,warning,error,debug That said, dialplan reload seems to show significantly fewer items than before. I've got loads of extensions, and it does indeed seem to reload them all, but only shows about 1/4 of them in the console when doing dialplan reload. Hmmm... Incidentally, just while I'm here, is there a particular reason that debug can only every be pushed higher when connecting to the console? it's always been the case since I started using Asterisk 3 years ago so I guess there's a reason, and I never questioned it before. I'm just curious! For example: asterisk -rvddd Core debug was 2 and is now 3. asterisk -rv Core debug was 3 and is now 4. asterisk -rvdd Core debug is still 4. asterisk -rvd Core debug is still 4. But it always respects "core set debug" in whichever direction of verbosity is required. Thanks again! On Sun, 29 Jul 2018 at 13:14, Richard Mudgett wrote: > > > > On Sat, Jul 28, 2018 at 1:10 PM, Jonathan H wrote: >> >> I've not needed to do a dialplan reload for a while, so I don't know >> exactly which version is stopped working, but on 15.5, I'm not seeing >> ANY debug info at any debug level. >> So I'm not really sure how to find mistakes in the dialplan. This is >> all I get... how do I enable this debug mode to see the previous >> behaviour? Thanks >> >> asterisk -rvd >> (enters console) >> dialplan reload >> Dialplan reloaded. >> [...] >> -- pbx_config successfully loaded 125 contexts (enable debug for >> details). > > > Many of those messages now go out as DEBUG level 1 messages. You would > need to enable those to go out to your console in logger.conf if they aren't > enabled. > Or you need to look at one of the logging files (like full) that has debug > messages > routed to it. > > https://issues.asterisk.org/jira/browse/ASTERISK-27084 is the issue that did > that > which went out in v15.3.0 and is mentioned in the CHANGES file: > > Core > -- > * During dialplan reload log messages are produced for each context, >extension and include. These messages are no longer printed by the >verbose loggers, they are now only logged as debug messages. > > > Richard > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan reload not showing debug info even with debug on (ast 15.5)
On Sat, Jul 28, 2018 at 1:10 PM, Jonathan H wrote: > I've not needed to do a dialplan reload for a while, so I don't know > exactly which version is stopped working, but on 15.5, I'm not seeing > ANY debug info at any debug level. > So I'm not really sure how to find mistakes in the dialplan. This is > all I get... how do I enable this debug mode to see the previous > behaviour? Thanks > > asterisk -rvd > (enters console) > dialplan reload > Dialplan reloaded. > [...] > -- pbx_config successfully loaded 125 contexts (enable debug for > details). > Many of those messages now go out as DEBUG level 1 messages. You would need to enable those to go out to your console in logger.conf if they aren't enabled. Or you need to look at one of the logging files (like full) that has debug messages routed to it. https://issues.asterisk.org/jira/browse/ASTERISK-27084 is the issue that did that which went out in v15.3.0 and is mentioned in the CHANGES file: Core -- * During dialplan reload log messages are produced for each context, extension and include. These messages are no longer printed by the verbose loggers, they are now only logged as debug messages. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dialplan reload not showing debug info even with debug on (ast 15.5)
I've not needed to do a dialplan reload for a while, so I don't know exactly which version is stopped working, but on 15.5, I'm not seeing ANY debug info at any debug level. So I'm not really sure how to find mistakes in the dialplan. This is all I get... how do I enable this debug mode to see the previous behaviour? Thanks asterisk -rvd (enters console) dialplan reload Dialplan reloaded. [...] -- pbx_config successfully loaded 125 contexts (enable debug for details). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan question: Variables in GoTo() ?
On Thu, 10 Mar 2016, A J Stiles wrote:
Can you use variables in the target of a GoTo() statement?
Yes. Here are a few examples from one of my dialplans:
; invalid template
[i](!)
exten = i,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = i,n,goto(${CONTEXT},s,1)
; look up (set) DNIS (DID) channel variables
exten = _x.,n, goto(lookup-dnis,${EXTEN},1)
; dispatch to selected application
exten = _[123456],n,
goto(${PRODUCT-${EXTEN}-APPLICATION},s,1)
This particular dialplan uses the invalid template in around 30 contexts
and 'goto(${CONTEXT},s,1)' about 15 times. Note that the last example
'nests' the variable expansion -- a variable within a variable.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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Re: [asterisk-users] Dialplan question: Variables in GoTo() ?
A J Stiles wrote:
I can't seem to find a definitive answer on this, and I really don't want to
risk breaking a production server to find out; so I am going to try asking this
here, and maybe anyone else in the same situation searching the archives
sometime in future will find the answer I get.
Can you use variables in the target of a GoTo() statement?
What I am specifically thinking of is this;
[from_some_source]
exten => s,1,AGI(look_up_stuff.agi,${CALLERID(num)},${EXTEN})
; this AGI script sets variables: next_context, next_ext, next_step
exten => s,n,GoTo(${next_context},${next_ext},${next_step})
Will this work? Does Asterisk evaluate expressions like this, or does it
expect literals?
It most certainly will work. It evaluates on use.
--
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Check us out at: www.digium.com & www.asterisk.org
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[asterisk-users] Dialplan question: Variables in GoTo() ?
I can't seem to find a definitive answer on this, and I really don't want to
risk breaking a production server to find out; so I am going to try asking this
here, and maybe anyone else in the same situation searching the archives
sometime in future will find the answer I get.
Can you use variables in the target of a GoTo() statement?
What I am specifically thinking of is this;
[from_some_source]
exten => s,1,AGI(look_up_stuff.agi,${CALLERID(num)},${EXTEN})
; this AGI script sets variables: next_context, next_ext, next_step
exten => s,n,GoTo(${next_context},${next_ext},${next_step})
Will this work? Does Asterisk evaluate expressions like this, or does it
expect literals?
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
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Re: [asterisk-users] dialplan contexts syntax and terminology
This one specifically http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics-SECT-3.html#asterisk-DP-Basics-SECT-3.1 On 22-Feb-2015 11:13 AM, thufir hawat.thu...@gmail.com wrote: On Sun, 22 Feb 2015 08:32:26 +0530, Mitul Limbani wrote: READ READ READ I know, I have the 4th edition and I've been reading it. Personally, I find it more general than specific, but I'll go back through that chapter, absolutely. thanks, Thufir -- _ -- 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 -- _ -- 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 contexts syntax and terminology
READ READ READ
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics.html
Regards,
Mitul Limbani,
Business Head,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422
On Sun, Feb 22, 2015 at 8:25 AM, thufir hawat.thu...@gmail.com wrote:
I'm looking into the dialplan specifics:
tleilax:~ #
tleilax:~ # cat /etc/asterisk/extensions.conf
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
TRUNK=DAHDI/r1; Trunk interface
TRUNKX=DAHDI/r2 ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKBINFONE=IAX2/111222:passw...@iax.binfone.com ; IAX trunk
interface
SIPtrunk=SIP/1234:passw...@sip.provider.net ; SIP trunk
#include extensions-vicidial.conf
Firstly, what language or format is this? Bash script?
the line #include ... what is this called? An include statement?
The [globals] -- what's the terminology for this? It's a context? And
a context is a logical separation in the dialplan? Is that, in any way,
analogous to a function or method?
Once you create your this logical separation, what's the syntax
surrounding invoking a specific context? For example:
tleilax:~ #
tleilax:~ # tail /etc/asterisk/extensions-vicidial.conf
[vicidial-auto]
exten = h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-
NODEBUG-${HANGUPCAUSE}-${DIALSTATUS}-${DIALEDTIME}-
${ANSWEREDTIME})
include = vicidial-auto-internal
include = vicidial-auto-phones
include = vicidial-auto-external
; END OF FILELast Forced System Reload: 2015-02-20 16:49:28
tleilax:~ #
when the above contexts are included, these contexts are declared within
the extensions-vicidial.conf, meaning that when they're declared, they're
not actually used/invoked/called **until** the actual include = foo
syntax?
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[asterisk-users] dialplan contexts syntax and terminology
I'm looking into the dialplan specifics:
tleilax:~ #
tleilax:~ # cat /etc/asterisk/extensions.conf
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
TRUNK=DAHDI/r1; Trunk interface
TRUNKX=DAHDI/r2 ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKBINFONE=IAX2/111222:passw...@iax.binfone.com ; IAX trunk
interface
SIPtrunk=SIP/1234:passw...@sip.provider.net ; SIP trunk
#include extensions-vicidial.conf
Firstly, what language or format is this? Bash script?
the line #include ... what is this called? An include statement?
The [globals] -- what's the terminology for this? It's a context? And
a context is a logical separation in the dialplan? Is that, in any way,
analogous to a function or method?
Once you create your this logical separation, what's the syntax
surrounding invoking a specific context? For example:
tleilax:~ #
tleilax:~ # tail /etc/asterisk/extensions-vicidial.conf
[vicidial-auto]
exten = h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-
NODEBUG-${HANGUPCAUSE}-${DIALSTATUS}-${DIALEDTIME}-
${ANSWEREDTIME})
include = vicidial-auto-internal
include = vicidial-auto-phones
include = vicidial-auto-external
; END OF FILELast Forced System Reload: 2015-02-20 16:49:28
tleilax:~ #
when the above contexts are included, these contexts are declared within
the extensions-vicidial.conf, meaning that when they're declared, they're
not actually used/invoked/called **until** the actual include = foo
syntax?
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Re: [asterisk-users] dialplan contexts syntax and terminology
On Sun, 22 Feb 2015 08:32:26 +0530, Mitul Limbani wrote: READ READ READ I know, I have the 4th edition and I've been reading it. Personally, I find it more general than specific, but I'll go back through that chapter, absolutely. thanks, Thufir -- _ -- 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 for receiving faxes on Asterisk
On 30/01/2015 1:25 PM, Simon Humbert wrote: Hi all, It looks like people commonly use this kind of dialplan when receiving faxes on Asterisk, with a jump to extension fax during the Wait() if a fax tone is detected: [start-here] exten = _X.,1,Answer() exten = _X.,n,Wait(n) exten = _X.,n,...do stuff... exten = _X.,n,Hangup() exten = fax,1,Goto(fax-rx,receive,1) [fax-rx] exten = receive,1,... exten = receive,n,...do stuff... exten = receive,n,ReceiveFAX() This is well suited in case Asterisk needs to receive both voice and fax calls. But what if Asterisk is only used to receive fax calls, can we start directly at the fax-rx context? I've heard that it's better to wait a few seconds before calling ReceiveFAX(), is it still necessary in case we don't actually need fax detection? If you don't have the need to detect the fax tone then I don't see any need to wait. You should disable the 'faxdetect' option in your peer otherwise it may attempt to redirect to the 'fax' extension upon detecting the fax signalling. Assuming you are using a SIP trunk to accept the call you could use in your sip.conf peer something like; context=fax-rx disallow=all allow=alaw,ulaw jbenable=no faxdetect=no directmedia=no callbackextension=receive t38pt_usertpsource=yes encryption=no Note, in this example I am using 'callbackextension' instead of 'register =', refer to the default sip.conf for further information. Larry. -- _ -- 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
[asterisk-users] Dialplan for receiving faxes on Asterisk
Hi all, It looks like people commonly use this kind of dialplan when receiving faxes on Asterisk, with a jump to extension fax during the Wait() if a fax tone is detected: [start-here] exten = _X.,1,Answer() exten = _X.,n,Wait(n) exten = _X.,n,...do stuff... exten = _X.,n,Hangup() exten = fax,1,Goto(fax-rx,receive,1) [fax-rx] exten = receive,1,... exten = receive,n,...do stuff... exten = receive,n,ReceiveFAX() This is well suited in case Asterisk needs to receive both voice and fax calls. But what if Asterisk is only used to receive fax calls, can we start directly at the fax-rx context? I've heard that it's better to wait a few seconds before calling ReceiveFAX(), is it still necessary in case we don't actually need fax detection? Simon -- _ -- 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
[asterisk-users] dialplan reload context
Hello, is it possible to reload just a context in stead of the whole dialplan ? I see this on the tracker : https://issues.asterisk.org/jira/browse/ASTERISK-19934 But is it possible in some Asterisk version ? Kind regards, Jonas. -- _ -- 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 reload context
Using current svn trunk, that option isn't available. It would appear that the patch from that issue did not get into the code. On Tue, Oct 28, 2014 at 10:22 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, is it possible to reload just a context in stead of the whole dialplan ? I see this on the tracker : https://issues.asterisk.org/jira/browse/ASTERISK-19934 But is it possible in some Asterisk version ? Kind regards, Jonas. -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 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 reload context
On Tue, Oct 28, 2014 at 10:54 AM, Scott Griepentrog sgriepent...@digium.com wrote: Using current svn trunk, that option isn't available. It would appear that the patch from that issue did not get into the code. On Tue, Oct 28, 2014 at 10:22 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, is it possible to reload just a context in stead of the whole dialplan ? I see this on the tracker : https://issues.asterisk.org/jira/browse/ASTERISK-19934 But is it possible in some Asterisk version ? The issue is still open, so yes, the patch hasn't been merged yet. If you'd like to help the process move forward faster with that improvement, there are a few things you can do: 1. You can test out the patch on the lastest Asterisk trunk, and provide feedback on the feature. 2. With the author's permission, you can post the patch for review to Review Board [1]. 3. Tests are always appreciated, particularly with new features that impact highly critical parts of the code, such as the PBX core. Writing tests for the feature will definitely help it get included faster [2]. [1] https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Test+Suite+Documentation Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
[asterisk-users] dialplan =how many concurrent calls
Hi guys. Does somebody knows how to get the concurrent calls from the dial plan? Or. How can i control not to run more than n simultaneus agi applications? Thanks in advance. rv -- _ -- 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 =how many concurrent calls
you can use GROUP and GROUP_COUNT
n,Set(GROUP()=aname)
n,GotoIf($[${GROUP_COUNT(aname)} 8]?${EXTEN},200)
200,Hangup
On Thu, Jul 10, 2014 at 3:24 PM, Rafael Visser visser.raf...@gmail.com
wrote:
Hi guys.
Does somebody knows how to get the concurrent calls from the dial plan?
Or.
How can i control not to run more than n simultaneus agi applications?
Thanks in advance.
rv
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Re: [asterisk-users] dialplan =how many concurrent calls
Works fine..
Thanks Asghar!
rv
2014-07-10 9:35 GMT-04:00 Asghar Mohammad asghar...@gmail.com:
you can use GROUP and GROUP_COUNT
n,Set(GROUP()=aname)
n,GotoIf($[${GROUP_COUNT(aname)} 8]?${EXTEN},200)
200,Hangup
On Thu, Jul 10, 2014 at 3:24 PM, Rafael Visser visser.raf...@gmail.com
wrote:
Hi guys.
Does somebody knows how to get the concurrent calls from the dial plan?
Or.
How can i control not to run more than n simultaneus agi applications?
Thanks in advance.
rv
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[asterisk-users] dialplan changes in middle of call
Recently, I made a change to our dialplan and reloaded Asterisk. To my surprise, the dialplan was reloaded for calls in progress. This caused a problem because some of the dialplan changes affected some loops and this caused an infinite loop. Is there a way to change this so that reloading Asterisk after a dialplan change affects new calls and not calls in progress? -H -- _ -- 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 changes in middle of call
Henry Fernandes wrote: Kia ora, Recently, I made a change to our dialplan and reloaded Asterisk. To my surprise, the dialplan was reloaded for calls in progress. This caused a problem because some of the dialplan changes affected some loops and this caused an infinite loop. Is there a way to change this so that reloading Asterisk after a dialplan change affects new calls and not calls in progress? Not without drastically changing the PBX core. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.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 to reach external SIP phone
Hi,
If (the other phone is also registered on same asterisk)
You just have to Dial(SIP/${EXTEN})
else
You need a trunk, or route the call to that
equipment(VoIP server) where the other phone(s) is/are registered so that
you can bridge both channels. Here you have to Dial(SIP/${EXTEN}@Trunk-IP)
or you can create a trunk in sip.conf to route call out, and the receiving
side should then route the call to the destination phone.
Regards
On Wed, Apr 2, 2014 at 8:11 AM, Meadows Hoa meadows_...@yahoo.com wrote:
If I have Asterisk setup with local SIP phones configured but need to call
a SIP phone which is not local but actually on another VLAN, what would the
dialplan need to look like?
Could the Asterisk dialplan directly call a SIP phone which is not a local
phone within its sip.conf and dialplan, if the Directory Number and IP is
known (or host name)?
Didn't plan on needing a SIP trunk so assuming this is possible anyway?
Would Asterisk have to actually go over a SIP trunk to another call
manager (which has that phone configured as a local phone)? Thinking there
has to be another way?
Any help would be appreciated. Thanks
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[asterisk-users] Dialplan to reach external SIP phone
If I have Asterisk setup with local SIP phones configured but need to call a SIP phone which is not local but actually on another VLAN, what would the dialplan need to look like? Could the Asterisk dialplan directly call a SIP phone which is not a local phone within its sip.conf and dialplan, if the Directory Number and IP is known (or host name)? Didn't plan on needing a SIP trunk so assuming this is possible anyway? Would Asterisk have to actually go over a SIP trunk to another call manager (which has that phone configured as a local phone)? Thinking there has to be another way? Any help would be appreciated. Thanks-- _ -- 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 MySQL inserted ID
On 20/08/13 17:48, Gergo Csibra wrote:
can I echo this variable ?
Like : exten = s,n,NoOp(${LAST_INSERT_ID()})
No, this is a mysql query, so:
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET C1=${ARG1},
C2=${ARG2}, timestamp=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
exten = s,n,MYSQL(Query resultid ${connid} SELECT LAST_INSERT_ID())
exten = s,n,NoOp(${resultid})
first is your original insert query, next you must read the
last_insert_id() mysql function with an other query, then you can
echo the resultid variable which contains the last inserted id.
I would be a bit concerned about doing this on a busy system. What would
happen if one call inserted a value, a second call inserted a value and
then the first call read the LAST_INSERT_ID? Would it get the wrong
value back?
If you do it in AGI then each query can have its own database connection
and so avoid this issue. If thats a problem use FastAGI and have a
daemon running and use transactions or another method to avoid the issue.
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[asterisk-users] Dialplan MySQL inserted ID
Hello,
how can I obtain the inserted ID after having inserted a row with
MySQL in the dialplan ?
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET
C1=${ARG1}, C2=${ARG2},
timestamp=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
I need to know the ID of the newly inserted row.
Kind regards,
Jonas.
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Re: [asterisk-users] Dialplan MySQL inserted ID
On 20/08/13 14:53, Jonas Kellens wrote:
Hello,
how can I obtain the inserted ID after having inserted a row with
MySQL in the dialplan ?
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET
C1=${ARG1}, C2=${ARG2},
timestamp=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
I need to know the ID of the newly inserted row.
You could write an AGI script in something like php or perl and get it
to write to the mysql database instead. It can then set a variable which
the dialplan can pick up.
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Re: [asterisk-users] Dialplan MySQL inserted ID
On Tuesday 20 August 2013, Jonas Kellens wrote:
Hello,
how can I obtain the inserted ID after having inserted a row with
MySQL in the dialplan ?
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET
C1=${ARG1}, C2=${ARG2},
timestamp=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
I need to know the ID of the newly inserted row.
Kind regards,
Jonas.
I'm not sure it's possible to do that using the simple MySQL interface
provided within the dialplan.
Why not write an AGI script in your favourite language (Perl, Python, PHP,
Java all have AGI and MySQL bindings) to perform the INSERT query for you?
You can supply values for C1 and C2 easily enough; and have your AGI script
return the insert ID in a channel variable. (You could also return another
channel variable indicating success or failure, if this is important.)
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Re: [asterisk-users] Dialplan MySQL inserted ID
Tuesday, August 20, 2013, 5:47:24 PM, Gareth wrote:
On 20/08/13 14:53, Jonas Kellens wrote:
Hello,
how can I obtain the inserted ID after having inserted a row with
MySQL in the dialplan ?
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET
C1=${ARG1}, C2=${ARG2},
timestamp=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
I need to know the ID of the newly inserted row.
You could write an AGI script in something like php or perl and get it
to write to the mysql database instead. It can then set a variable which
the dialplan can pick up.
meh...
SELECT LAST_INSERT_ID()
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Re: [asterisk-users] Dialplan MySQL inserted ID
On 08/20/2013 06:03 PM, Gergo Csibra wrote:
Tuesday, August 20, 2013, 5:47:24 PM, Gareth wrote:
On 20/08/13 14:53, Jonas Kellens wrote:
Hello,
how can I obtain the inserted ID after having inserted a row with
MySQL in the dialplan ?
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET
C1=${ARG1}, C2=${ARG2},
timestamp=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
I need to know the ID of the newly inserted row.
You could write an AGI script in something like php or perl and get it
to write to the mysql database instead. It can then set a variable which
the dialplan can pick up.
meh...
SELECT LAST_INSERT_ID()
Hello,
can I echo this variable ?
Like : exten = s,n,NoOp(${LAST_INSERT_ID()})
Kind regards,
Jonas.
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Re: [asterisk-users] Dialplan MySQL inserted ID
On 20/8/13 5:00 pm, A J Stiles wrote:
Why not write an AGI script in your favourite language (Perl, Python, PHP,
Java all have AGI and MySQL bindings) to perform the INSERT query for you?
+1. It would also give you somewhere to perform sanity checks on your
${ARGS} to avoid SQL injection attacks...
Kind regards,
Chris
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Re: [asterisk-users] Dialplan MySQL inserted ID
Tuesday, August 20, 2013, 6:08:19 PM, Jonas wrote:
On 08/20/2013 06:03 PM, Gergo Csibra wrote:
Tuesday, August 20, 2013, 5:47:24 PM, Gareth wrote:
On 20/08/13 14:53, Jonas Kellens wrote:
Hello,
how can I obtain the inserted ID after having inserted a row with
MySQL in the dialplan ?
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET
C1=${ARG1}, C2=${ARG2},
timestamp=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
I need to know the ID of the newly inserted row.
You could write an AGI script in something like php or perl and get it
to write to the mysql database instead. It can then set a variable which
the dialplan can pick up.
meh...
SELECT LAST_INSERT_ID()
Hello,
can I echo this variable ?
Like : exten = s,n,NoOp(${LAST_INSERT_ID()})
No, this is a mysql query, so:
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET
C1=${ARG1}, C2=${ARG2},
timestamp=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
exten = s,n,MYSQL(Query resultid ${connid} SELECT LAST_INSERT_ID())
exten = s,n,NoOp(${resultid})
first is your original insert query, next you must read the
last_insert_id() mysql function with an other query, then you can
echo the resultid variable which contains the last inserted id.
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Re: [asterisk-users] + dialplan
Hello Adam,
Thank you very much for your info.
Regards,
Jonson.
On Tue, Jun 11, 2013 at 12:34 AM, ad...@3a.hu wrote:
Hi,
On 06/10/2013 22:26, Jonson Player wrote:
Some users of main use + instead of 00 for international dial. Is there
any solution for this problem?
swap the + sign to double zeros if your provider can't handle it
; normal 00 prefix
exten = _00ZZXXX.,1,Macro(**beforealldials)
exten = _00ZZXXX.,n,Dial(SIP/${**EXTEN}@${OUTGOING_LINE})
exten = _00ZZXXX.,n,Hangup()
; swap + prefix to 00
exten = _+ZZXXX.,1,Macro(**beforealldials)
exten = _+ZZXXX.,n,Dial(SIP/00${**EXTEN:1}@${OUTGOING_LINE})
exten = _+ZZXXX.,n,Hangup()
regards
adam
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[asterisk-users] + dialplan
Hello guys, I looking for some dial plan which can mach on +xxx numbers instead of 00xxx numbers. Some users of main use + instead of 00 for international dial. Is there any solution for this problem? As far as i readed in asterisk is some kind of replacement of characters in dial plan command. Could i use that for archiving this option? Thank you for help. Jonson. -- _ -- 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
Hi,
On 06/10/2013 22:26, Jonson Player wrote:
Some users of main use + instead of 00 for international dial. Is there
any solution for this problem?
swap the + sign to double zeros if your provider can't handle it
; normal 00 prefix
exten = _00ZZXXX.,1,Macro(beforealldials)
exten = _00ZZXXX.,n,Dial(SIP/${EXTEN}@${OUTGOING_LINE})
exten = _00ZZXXX.,n,Hangup()
; swap + prefix to 00
exten = _+ZZXXX.,1,Macro(beforealldials)
exten = _+ZZXXX.,n,Dial(SIP/00${EXTEN:1}@${OUTGOING_LINE})
exten = _+ZZXXX.,n,Hangup()
regards
adam
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[asterisk-users] Dialplan reload not reloading everything
Good morning, We recently fell back to the most recent build of asterisk 1.8 down from 11.3 and I believe we've crossed some sort of limit for 1.8. Our dialplan is 515723 entries long with 6263 distinct contexts. Both are loaded realtime via odbc (mysql). Previously at the end of a dialplan reload we would get a summary of how long it took to reload everything. Now it just shows the last line that it loaded as seen below: -- Registered extension context '4959_5095_0'; registrar: pbx_config -- Including switch 'realtime/@' in context '4959_5095_0' -- Registered extension context '4960_5096_0'; registrar: pbx_config -- Including switch 'realtime/@' in context '4960_5096_0' asterisk*CLI I've tried turning on debug and there's no extra information. I tried running it from the command line with -rx and it says dialplan reloaded. Yet a bunch of the newest contexts aren't recognized (asterisk reports that they don't exist then call dies). I've confirmed they're in the database. The context it ends with is not always the same. Sometimes it's in the 3000 range and sometimes it's in the 4000s. Has anyone else seen this? Is there a maximum string length for the contexts or something that could be causing this? -- _ -- 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 reload not reloading everything
- Original Message - From: Brandon Mackie bmac...@awktane.com We recently fell back to the most recent build of asterisk 1.8 down from 11.3 and I believe we’ve crossed some sort of limit for 1.8. Our dialplan is 515723 entries long with 6263 distinct contexts. Both are loaded realtime via odbc (mysql). That's a big dialplan! snip I’ve tried turning on debug and there’s no extra information. snip How did you enable debug? Did you follow the directions here? https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information If you can pastebin a log showing the end of the reload with WARNING,ERROR,NOTICE, plus VERBOSE and DEBUG turned up (try 5 or higher) then maybe there will be something interesting.. -- Rusty Newton OS Community Support Manager | Digium, Inc. | www.digium.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
[asterisk-users] Dialplan / check / tool
Hi, I am wondering, if there is any tool available, which performs a check for suspicious entries in the dialplan. For example a non existing AGI-Script or a double assigned extension ike that: [context] exten = *100*,1,AGI(test_app.pl) ... exten = 190,1,Answer() ... exten = *100*,1,AGI(never_reached.pl) ... A normal dialplan reload command would echo no warning or something similair. Best regards -Thorsten- -- _ -- 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 / check / tool
On Mon, Feb 18, 2013 at 8:11 AM, Thorsten Göllner t...@ovm-group.com wrote: A normal dialplan reload command would echo no warning or something similair. The duplicated extension will cause an error. Something like cannot add extension in line X because it already exists. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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 / check / tool
A normal dialplan reload command would echo no warning or something similair. Normally I would see these being logged to /var/log/asterisk/messages with a stock Asterisk install. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 / check / tool
On Mon, 18 Feb 2013, Thorsten Göllner wrote:
I am wondering, if there is any tool available, which performs a check
for suspicious entries in the dialplan. For example a non existing
AGI-Script...
I'm just a 1.2 Luddite, but none that I know of.
Please feel free to write one. Here are a few features that would be
helpful:
) Parentheses, bracket, brace, quote, and double-quote matching.
) Parse /etc/init.d/asterisk to see if -C is used; parse asterisk.conf to
see if astagidir is defined; take note of the username used to start
Asterisk.
) Parse extensions.conf for global variables to cover the use case of:
exten = *,n, agi(${SOME-VARIABLE}/foo)
) Check the permissions of the AGI's path relative to the username and
group that starts Asterisk. Warn if silly permissions like 777 are found.
) If the AGI is an interpreted script (Bash, Perl, PHP, Python, etc.)
instead of a compiled executable (C, Fortran, Cobol, assembler, etc.)
ensure that the interpreter is present and functional (maybe something
like 'interpreter --version').
) Detect dialplan 'fall-through.'
) Detect 'gaps' in priorities. Note that priorities do not need to be
contiguous or even specified in sequential order.
) Have a command line parameter to specify which version of Asterisk to
check compliance against. This would be a great 'desk check' before
migrating from 1.2 to 11 :)
) Detect global and channel variables defined, but not used.
) Detect global and channel variables used before defined.
) Detect missing goto targets.
Feel free to implement a subset of the above for your initial release :)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
_
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Re: [asterisk-users] Dialplan / check / tool
On Mon, Feb 18, 2013 at 10:54 AM, Steve Edwards asterisk@sedwards.comwrote: ) If the AGI is [...] a compiled executable (C, Fortran, Cobol, assembler, etc.) I'd like to see an AGI written using Fortran or Cobol. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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 / check / tool
On 18 Feb 2013, at 17:03, Christopher Harrington wrote: On Mon, Feb 18, 2013 at 10:54 AM, Steve Edwards asterisk@sedwards.com wrote: ) If the AGI is [...] a compiled executable (C, Fortran, Cobol, assembler, etc.) I'd like to see an AGI written using Fortran or Cobol. Don't tempt me ;) S-- _ -- 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 / check / tool
On Mon, 18 Feb 2013, Steven Howes wrote: On 18 Feb 2013, at 17:03, Christopher Harrington wrote: On Mon, Feb 18, 2013 at 10:54 AM, Steve Edwards asterisk@sedwards.com wrote: ) If the AGI is [...] a compiled executable (C, Fortran, Cobol, assembler, etc.) I'd like to see an AGI written using Fortran or Cobol. Don't tempt me ;) I ate the apple before Eve... program jwb print*, 'verbose Welcome to 1957' read(*,*) end program jwb It violates the AGI protocol (doesn't read the AGI variables) but it does execute correctly. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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 - working out when users answer
Hey Satish,
I've worked this out. I'm sorry, you were completely right and the context is
fine. I was testing without answering the call, so the Dial was never
connected! Doh!
Thanks heaps for your help, it's all working perfectly.
Cheers,
Andrew
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Barot
Sent: Tuesday, 8 January 2013 12:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialplan - working out when users answer
HI Andrew,
Show your queuecontrol context. You should have extension s with priority 1 in
this context.
--Satish Barot
On Mon, Jan 7, 2013 at 12:08 PM, Andrew White
and...@computersforall.com.aumailto:and...@computersforall.com.au wrote:
Hi Satish,
Thanks for your response - sorry on the slow reply.
So I've tried the following in the dialplan:
exten =
direct,n,Dial(${QUEUEEXTS},${RINGTIME},U(queueControl,direct^CONNECTED))
This has a very strange behavior - the NoOp that is in
queueControl,direct,n(CONNECTED) does not show up, however I get the following:
[2013-01-07 17:31:39] ERROR[19135]: app_stack.c:420 gosub_exec: Attempt to
reach a non-existent destination for gosub: (Context:queueControl, Extension:s,
Priority:1)
I've also tried with a macro:
exten = direct,n,Dial(${QUEUEEXTS},${RINGTIME},M(inboundconnected))
[macro-inboundconnected]
exten = s,1,NoOp(Inbound connected!)
It definitely seems like it's being called, but again no NoOp:
-- Executing [direct@queueControl:11] Dial(SIP/1000-47f1,
SIP/1000,20,M(inboundconnected)) in new stack
I would expect some kind of error if I was doing this wrong - have I missed
something?
Thanks for your or anyone elses help in advance!
Andrew
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialplan - working out when users answer
On Wed, Dec 19, 2012 at 12:44 PM, Andrew White
and...@computersforall.com.aumailto:and...@computersforall.com.au wrote:
Hi Satish/list,
Looks like I spoke to soon.
I have the following in my dialplan:
Dial(${QUEUEEXTS},${RINGTIME},U(queueControl^direct^CONNECTED))
And after confirming with a dialplan show it was definitely in there, I
continued to get this:
ERROR[28167]: app_stack.c:420 gosub_exec: Attempt to reach a non-existent
destination for gosub: (Context:queueControl, Extension:s, Priority:1)
I can't quite work out why it would be trying to s/1 instead of
direct/CONNECTED =/.
Any ideas?
Thanks!
In your case, direct and CONNECTED have to be arguments and not the extension
and priority value respectively. Calling Subroutine from dial will always start
execution with extension s and priority 1.
See the link for more information, Arguments are passed to subroutine using ^
as a delimiter.
--Satish Barot
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Dialplan - working out when users answer
Thanks Satish, fantastic advice. I didn't even think to look into the dial
options - doh!
Thanks very much,
Andrew
On Wed, Dec 19, 2012 at 10:53 AM, Andrew White
and...@computersforall.com.aumailto:and...@computersforall.com.au wrote:
Hey guys,
I've got a part of my dialplan that dials multiple people:
exten = direct,n,Dial(${QUEUEEXTS},${RINGTIME})
Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g.
SIP/100SIP/101SIP/105 etc
This works great, however I want to see if I can find a way to work out (and
run an AGI script) when the call is picked up by someone.
Thanks all!
Option M or U of Dial application would help you do this.
https://wiki.asterisk.org/wiki/display/AST/Application_Dial.
--Satish Barot
--
_
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Re: [asterisk-users] Dialplan - working out when users answer
HI Andrew,
Show your queuecontrol context. You should have extension s with priority
1 in this context.
--Satish Barot
On Mon, Jan 7, 2013 at 12:08 PM, Andrew White and...@computersforall.com.au
wrote:
Hi Satish,
** **
Thanks for your response – sorry on the slow reply.
** **
So I’ve tried the following in the dialplan:
** **
exten =
direct,n,Dial(${QUEUEEXTS},${RINGTIME},U(queueControl,direct^CONNECTED))**
**
** **
This has a very strange behavior – the NoOp that is in
queueControl,direct,n(CONNECTED) does not show up, however I get the
following:
** **
*[2013-01-07 17:31:39] ERROR[19135]: app_stack.c:420 gosub_exec: Attempt
to reach a non-existent destination for gosub: (Context:queueControl,
Extension:s, Priority:1)*
** **
I’ve also tried with a macro:
** **
exten = direct,n,Dial(${QUEUEEXTS},${RINGTIME},M(inboundconnected))
[macro-inboundconnected]
exten = s,1,NoOp(Inbound connected!)
** **
It definitely seems like it’s being called, but again no NoOp:
** **
*-- Executing [direct@queueControl:11] Dial(SIP/1000-47f1,
SIP/1000,20,M(inboundconnected)) in new stack*
* *
I would expect some kind of error if I was doing this wrong – have I
missed something?
** **
Thanks for your or anyone elses help in advance!
** **
Andrew
* *
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Dialplan - working out when users answer**
**
** **
On Wed, Dec 19, 2012 at 12:44 PM, Andrew White
and...@computersforall.com.au wrote:
Hi Satish/list,
Looks like I spoke to soon.
I have the following in my dialplan:
*Dial(${QUEUEEXTS},${RINGTIME},U(queueControl^direct^CONNECTED))*
And after confirming with a “dialplan show” it was definitely in there, I
continued to get this:
*ERROR[28167]: app_stack.c:420 gosub_exec: Attempt to reach a
non-existent destination for gosub: (Context:queueControl, Extension:s,
Priority:1)*
* *
I can’t quite work out why it would be trying to s/1 instead of
direct/CONNECTED =/.
Any ideas?
Thanks!
In your case, direct and CONNECTED have to be arguments and not the
extension and priority value respectively. Calling Subroutine from dial
will always start execution with extension s and priority 1.
See the link for more information, Arguments are passed to subroutine
using ^ as a delimiter.
** **
--Satish Barot
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* RE: [asterisk-users] Dialplan - working out when users answer**
**
Thanks Satish, fantastic advice. I didn’t even think to look into the dial
options – doh!
Thanks very much,
Andrew
On Wed, Dec 19, 2012 at 10:53 AM, Andrew White
and...@computersforall.com.au wrote:
Hey guys,
I’ve got a part of my dialplan that dials multiple people:
*exten = direct,n,Dial(${QUEUEEXTS},${RINGTIME})
*Multiple extensions are in the ${QUEUEEXTS} from an external script –
e.g. SIP/100SIP/101SIP/105 etc
This works great, however I want to see if I can find a way to work out
(and run an AGI script) when the call is picked up by someone.
Thanks all!
Option M or U of Dial application would help you do this.
https://wiki.asterisk.org/wiki/display/AST/Application_Dial.
--Satish Barot
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan - working out when users answer
Hi Satish,
Thanks for your response - sorry on the slow reply.
So I've tried the following in the dialplan:
exten =
direct,n,Dial(${QUEUEEXTS},${RINGTIME},U(queueControl,direct^CONNECTED))
This has a very strange behavior - the NoOp that is in
queueControl,direct,n(CONNECTED) does not show up, however I get the following:
[2013-01-07 17:31:39] ERROR[19135]: app_stack.c:420 gosub_exec: Attempt to
reach a non-existent destination for gosub: (Context:queueControl, Extension:s,
Priority:1)
I've also tried with a macro:
exten = direct,n,Dial(${QUEUEEXTS},${RINGTIME},M(inboundconnected))
[macro-inboundconnected]
exten = s,1,NoOp(Inbound connected!)
It definitely seems like it's being called, but again no NoOp:
-- Executing [direct@queueControl:11] Dial(SIP/1000-47f1,
SIP/1000,20,M(inboundconnected)) in new stack
I would expect some kind of error if I was doing this wrong - have I missed
something?
Thanks for your or anyone elses help in advance!
Andrew
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Barot
Sent: Wednesday, 19 December 2012 7:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialplan - working out when users answer
On Wed, Dec 19, 2012 at 12:44 PM, Andrew White
and...@computersforall.com.aumailto:and...@computersforall.com.au wrote:
Hi Satish/list,
Looks like I spoke to soon.
I have the following in my dialplan:
Dial(${QUEUEEXTS},${RINGTIME},U(queueControl^direct^CONNECTED))
And after confirming with a dialplan show it was definitely in there, I
continued to get this:
ERROR[28167]: app_stack.c:420 gosub_exec: Attempt to reach a non-existent
destination for gosub: (Context:queueControl, Extension:s, Priority:1)
I can't quite work out why it would be trying to s/1 instead of
direct/CONNECTED =/.
Any ideas?
Thanks!
In your case, direct and CONNECTED have to be arguments and not the extension
and priority value respectively. Calling Subroutine from dial will always start
execution with extension s and priority 1.
See the link for more information, Arguments are passed to subroutine using ^
as a delimiter.
--Satish Barot
From: Andrew White
Sent: Wednesday, 19 December 2012 5:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Dialplan - working out when users answer
Thanks Satish, fantastic advice. I didn't even think to look into the dial
options - doh!
Thanks very much,
Andrew
From:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Barot
Sent: Wednesday, 19 December 2012 4:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialplan - working out when users answer
On Wed, Dec 19, 2012 at 10:53 AM, Andrew White
and...@computersforall.com.aumailto:and...@computersforall.com.au wrote:
Hey guys,
I've got a part of my dialplan that dials multiple people:
exten = direct,n,Dial(${QUEUEEXTS},${RINGTIME})
Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g.
SIP/100SIP/101SIP/105 etc
This works great, however I want to see if I can find a way to work out (and
run an AGI script) when the call is picked up by someone.
Thanks all!
Option M or U of Dial application would help you do this.
https://wiki.asterisk.org/wiki/display/AST/Application_Dial.
--Satish Barot
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
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_
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Re: [asterisk-users] Dialplan - working out when users answer
On Wed, Dec 19, 2012 at 12:44 PM, Andrew White
and...@computersforall.com.au wrote:
Hi Satish/list,
** **
Looks like I spoke to soon.
** **
I have the following in my dialplan:
** **
*Dial(${QUEUEEXTS},${RINGTIME},U(queueControl^direct^CONNECTED))*
** **
And after confirming with a “dialplan show” it was definitely in there, I
continued to get this:
** **
*ERROR[28167]: app_stack.c:420 gosub_exec: Attempt to reach a
non-existent destination for gosub: (Context:queueControl, Extension:s,
Priority:1)*
* *
I can’t quite work out why it would be trying to s/1 instead of
direct/CONNECTED =/.
** **
Any ideas?
** **
Thanks!
In your case, direct and CONNECTED have to be arguments and not the
extension and priority value respectively. Calling Subroutine from dial
will always start execution with extension s and priority 1.
See the link for more information, Arguments are passed to subroutine using
^ as a delimiter.
--Satish Barot
** **
*From:* Andrew White
*Sent:* Wednesday, 19 December 2012 5:58 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* RE: [asterisk-users] Dialplan - working out when users answer**
**
** **
Thanks Satish, fantastic advice. I didn’t even think to look into the dial
options – doh!
** **
Thanks very much,
** **
Andrew
** **
*From:* asterisk-users-boun...@lists.digium.com [
mailto:asterisk-users-boun...@lists.digium.comasterisk-users-boun...@lists.digium.com]
*On Behalf Of *Satish Barot
*Sent:* Wednesday, 19 December 2012 4:40 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Dialplan - working out when users answer**
**
** **
On Wed, Dec 19, 2012 at 10:53 AM, Andrew White
and...@computersforall.com.au wrote:
Hey guys,
I’ve got a part of my dialplan that dials multiple people:
*exten = direct,n,Dial(${QUEUEEXTS},${RINGTIME})
*Multiple extensions are in the ${QUEUEEXTS} from an external script –
e.g. SIP/100SIP/101SIP/105 etc
This works great, however I want to see if I can find a way to work out
(and run an AGI script) when the call is picked up by someone.
Thanks all!
Option M or U of Dial application would help you do this.
https://wiki.asterisk.org/wiki/display/AST/Application_Dial.
** **
--Satish Barot
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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_
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[asterisk-users] Dialplan - working out when users answer
Hey guys,
I've got a part of my dialplan that dials multiple people:
exten = direct,n,Dial(${QUEUEEXTS},${RINGTIME})
Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g.
SIP/100SIP/101SIP/105 etc
This works great, however I want to see if I can find a way to work out (and
run an AGI script) when the call is picked up by someone.
Thanks all!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan - working out when users answer
On Wed, Dec 19, 2012 at 10:53 AM, Andrew White
and...@computersforall.com.au wrote:
Hey guys,
** **
I’ve got a part of my dialplan that dials multiple people:
** **
*exten = direct,n,Dial(${QUEUEEXTS},${RINGTIME})
*Multiple extensions are in the ${QUEUEEXTS} from an external script –
e.g. SIP/100SIP/101SIP/105 etc
** **
This works great, however I want to see if I can find a way to work out
(and run an AGI script) when the call is picked up by someone.
** **
Thanks all!
** **
Option M or U of Dial application would help you do this.
https://wiki.asterisk.org/wiki/display/AST/Application_Dial.
--Satish Barot
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan - working out when users answer
Thanks Satish, fantastic advice. I didn't even think to look into the dial
options - doh!
Thanks very much,
Andrew
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Barot
Sent: Wednesday, 19 December 2012 4:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialplan - working out when users answer
On Wed, Dec 19, 2012 at 10:53 AM, Andrew White
and...@computersforall.com.aumailto:and...@computersforall.com.au wrote:
Hey guys,
I've got a part of my dialplan that dials multiple people:
exten = direct,n,Dial(${QUEUEEXTS},${RINGTIME})
Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g.
SIP/100SIP/101SIP/105 etc
This works great, however I want to see if I can find a way to work out (and
run an AGI script) when the call is picked up by someone.
Thanks all!
Option M or U of Dial application would help you do this.
https://wiki.asterisk.org/wiki/display/AST/Application_Dial.
--Satish Barot
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Re: [asterisk-users] Dialplan - working out when users answer
Hi Satish/list,
Looks like I spoke to soon.
I have the following in my dialplan:
Dial(${QUEUEEXTS},${RINGTIME},U(queueControl^direct^CONNECTED))
And after confirming with a dialplan show it was definitely in there, I
continued to get this:
ERROR[28167]: app_stack.c:420 gosub_exec: Attempt to reach a non-existent
destination for gosub: (Context:queueControl, Extension:s, Priority:1)
I can't quite work out why it would be trying to s/1 instead of
direct/CONNECTED =/.
Any ideas?
Thanks!
From: Andrew White
Sent: Wednesday, 19 December 2012 5:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Dialplan - working out when users answer
Thanks Satish, fantastic advice. I didn't even think to look into the dial
options - doh!
Thanks very much,
Andrew
From:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Barot
Sent: Wednesday, 19 December 2012 4:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialplan - working out when users answer
On Wed, Dec 19, 2012 at 10:53 AM, Andrew White
and...@computersforall.com.aumailto:and...@computersforall.com.au wrote:
Hey guys,
I've got a part of my dialplan that dials multiple people:
exten = direct,n,Dial(${QUEUEEXTS},${RINGTIME})
Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g.
SIP/100SIP/101SIP/105 etc
This works great, however I want to see if I can find a way to work out (and
run an AGI script) when the call is picked up by someone.
Thanks all!
Option M or U of Dial application would help you do this.
https://wiki.asterisk.org/wiki/display/AST/Application_Dial.
--Satish Barot
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Re: [asterisk-users] dialplan reloading
Hi Jerry, From the Asterisk CLI, enter the command core restart when convenient, this command will restart asterisk only when there is no incoming call, and when it will close all outgoing calls. With a restart of asterisk should reload all the information: extensions, sip, agi, iax, voicemail, etc. Danilo Il 01/11/12 23:30, Jerry Geis ha scritto: If I issue a dialplan reload and some AGI starts as its reloading and directs something into the diaplan that is still reloading what happens I presume my context is not there? What I see is the diaplan is messed up somehow and I goto the default context then after that it is messaged up until I stop and restart. How do i prevent this from happening? 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 reloading
Hi Jerry, From the Asterisk CLI, enter the command core restart when convenient, this command will restart asterisk only when there is no incoming call, and when it will close all outgoing calls. With a restart of asterisk should reload all the information: extensions, sip, agi, iax, voicemail, etc. Danilo Danilo, Ok - but then what if a call comes in while it decides to reload - or if an AGI is started while it decides to reload - Sure there is nothing happening at that moment - but lets say right after it decides that its convenient and before its done - something gets started. What to do about that? 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 reloading
On Fri, 2012-11-02 at 06:25 -0400, Jerry Geis wrote: Hi Jerry, From the Asterisk CLI, enter the command core restart when convenient, this command will restart asterisk only when there is no incoming call, and when it will close all outgoing calls. With a restart of asterisk should reload all the information: extensions, sip, agi, iax, voicemail, etc. Danilo Danilo, Ok - but then what if a call comes in while it decides to reload - or if an AGI is started while it decides to reload - Sure there is nothing happening at that moment - but lets say right after it decides that its convenient and before its done - something gets started. What to do about that? Jerry -- Sorry to step in here but I think the 2 of you are talking at cropp purposes I initial query was about a dialplan reload, not an asterisk restart. Jerry, how long does your system take to perform a dialplan reload? surely it is under a second. If you look in the logs, at the end of any dialplan reload (well in 1.8 at least) you will get some time stats such as below [2012-10-29 00:10:03] VERBOSE[19096] pbx.c: -- Time to scan old dialplan and merge leftovers back into the new: 0.41 sec [2012-10-29 00:10:03] VERBOSE[19096] pbx.c: -- Time to restore hints and swap in new dialplan: 0.03 sec [2012-10-29 00:10:03] VERBOSE[19096] pbx.c: -- Time to delete the old dialplan: 0.04 sec [2012-10-29 00:10:03] VERBOSE[19096] pbx.c: -- Total time merge_contexts_delete: 0.48 sec As you can see in this example it takes under a ten thousandth of a second. Is that something to really be concerned about? Regards Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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 reloading
Sorry to step in here but I think the 2 of you are talking at cropp purposes I initial query was about a dialplan reload, not an asterisk restart. Jerry, how long does your system take to perform a dialplan reload? surely it is under a second. If you look in the logs, at the end of any dialplan reload (well in 1.8 at least) you will get some time stats such as below [2012-10-29 00:10:03] VERBOSE[19096] pbx.c: -- Time to scan old dialplan and merge leftovers back into the new: 0.41 sec [2012-10-29 00:10:03] VERBOSE[19096] pbx.c: -- Time to restore hints and swap in new dialplan: 0.03 sec [2012-10-29 00:10:03] VERBOSE[19096] pbx.c: -- Time to delete the old dialplan: 0.04 sec [2012-10-29 00:10:03] VERBOSE[19096] pbx.c: -- Total time merge_contexts_delete: 0.48 sec As you can see in this example it takes under a ten thousandth of a second. Is that something to really be concerned about? Regards Ish Actually my mistake - looks like based on my code certain things happen and I issue two dialplan reload commands. So the second is killing the first. Then asterisk looses information. So certainly I should not be doing that - but I'm surprised asterisk lets another reload happen before the first is complete. 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 reloading
Jerry Geis wrote: Actually my mistake - looks like based on my code certain things happen and I issue two dialplan reload commands. So the second is killing the first. Then asterisk looses information. So certainly I should not be doing that - but I'm surprised asterisk lets another reload happen before the first is complete. What version of Asterisk are you running? There was an issue found in February where this exact behavior could occur, two dialplan reload commands would clobber each other. It was also resolved back then in all supported branches (1.8, 10, and trunk). http://lists.digium.com/pipermail/asterisk-commits/2012-February/053537.html For the 1.8 fix if you are curious. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.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 reloading
What version of Asterisk are you running? There was an issue found in February where this exact behavior could occur, two dialplan reload commands would clobber each other. It was also resolved back then in all supported branches (1.8, 10, and trunk). http://lists.digium.com/pipermail/asterisk-commits/2012-February/053537.html For the 1.8 fix if you are curious. Josh, I am running 1.4.43, planning on switching to 11 but have not got there... 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
[asterisk-users] dialplan reloading
If I issue a dialplan reload and some AGI starts as its reloading and directs something into the diaplan that is still reloading what happens I presume my context is not there? What I see is the diaplan is messed up somehow and I goto the default context then after that it is messaged up until I stop and restart. How do i prevent this from happening? 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 problem : not including context
On 01/13/2012 06:58 PM, Administrator TOOTAI wrote:
Le 13/01/2012 14:32, Jonas Kellens a écrit :
On 01/13/2012 02:23 PM, Doug Lytle wrote:
Jonas Kellens wrote:
I have the following in dialplan :
[TrunkAccounts]
dialplan show TrunkAccounts
Make sure the sort order is what you're expecting.
Doug
Hello,
The order is correct for as far as I'm sure.
[TrunkAccounts]
exten = 32380837,1,GoTo(01,32380837,1)
exten = 32380838,1,GoTo(01,32380838,1)
exten = 32380839,1,GoTo(01,32380839,1)
[CheckOnNet]
include = TrunkAccounts
exten = _321[0-3],1,GoTo(context1,${EXTEN},1)
exten = 3214,1,GoTo(context2, ${EXTEN} ,1)
exten = _.,1,NoOp()
exten = _.,n,Return()
Are you sure about your _. exten? Typo in the mail? It means 9
and more digits but your extensions are 8 digits ...
Include are always treated *after* context command. If _. is
right, something is wrong with Asterisk as it should treat
TrunkAccounts. If _XXX. (8 digits or more) is what you have in
yourdialplan, than the behavior of Asterisk is OK
Try
[TrunkAccounts]
exten = 32380837,1,GoTo(01,32380837,1)
exten = 32380838,1,GoTo(01,32380838,1)
exten = 32380839,1,GoTo(01,32380839,1)
[TrunkNotTreated]
exten = _.,1,NoOp()
exten = _.,n,Return()
[CheckOnNet]
include = TrunkAccounts
include = TrunkNotTreated
exten = _321[0-3],1,GoTo(context1,${EXTEN},1)
exten = 3214,1,GoTo(context2, ${EXTEN} ,1)
[...]
Hello,
I confirm that this is working for me !
Thanks !
Jonas.
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[asterisk-users] dialplan problem : not including context
Hello, I have the following in dialplan : [TrunkAccounts] exten = 32380837,1,GoTo(01,32380837,1) exten = 32380838,1,GoTo(01,32380838,1) exten = 32380839,1,GoTo(01,32380839,1) [CheckOnNet] include = TrunkAccounts But when a call for 32380837 enters CheckOnNet, it is not found. How come ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan problem : not including context
Jonas Kellens wrote: I have the following in dialplan : [TrunkAccounts] dialplan show TrunkAccounts Make sure the sort order is what you're expecting. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan problem : not including context
On 01/13/2012 02:23 PM, Doug Lytle wrote:
Jonas Kellens wrote:
I have the following in dialplan :
[TrunkAccounts]
dialplan show TrunkAccounts
Make sure the sort order is what you're expecting.
Doug
Hello,
The order is correct for as far as I'm sure.
[TrunkAccounts]
exten = 32380837,1,GoTo(01,32380837,1)
exten = 32380838,1,GoTo(01,32380838,1)
exten = 32380839,1,GoTo(01,32380839,1)
[CheckOnNet]
include = TrunkAccounts
exten = _321[0-3],1,GoTo(context1,${EXTEN},1)
exten = 3214,1,GoTo(context2,${EXTEN},1)
exten = _.,1,NoOp()
exten = _.,n,Return()
This is what I see on the CLI :
/[Jan 13 14:30:01] -- Executing [s@macro-uit789:47]
Gosub(SIP/yoc1-5a3c, CheckOnNet,//32380837,1) in new stack
[Jan 13 14:30:01] -- Executing [32380837@CheckOnNet:1]
NoOp(SIP/yoc1-5a3c, ) in new stack
[Jan 13 14:30:01] -- Executing [32380837//@CheckOnNet:2]
Return(SIP/yoc1-5a3c, ) in new stack/
So the context TrunkAccounts is not included.
Do you know why ?
Jonas.
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Re: [asterisk-users] dialplan problem : not including context
On 1/13/12 2:32 PM, Jonas Kellens wrote: So the context TrunkAccounts is not included. Do you know why ? Does reloading the dialplan (dialplan reload) give any useful output relating to these two contexts? -- Andreas Sikkema -- _ -- 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 problem : not including context
On 01/13/2012 02:37 PM, Andreas Sikkema wrote: On 1/13/12 2:32 PM, Jonas Kellens wrote: So the context TrunkAccounts is not included. Do you know why ? Does reloading the dialplan (dialplan reload) give any useful output relating to these two contexts? I include this context in 2 other contexts : [Jan 13 14:19:12] VERBOSE[4220] config.c: [Jan 13 14:19:12] == Parsing '/etc/asterisk/extensions.conf': [Jan 13 14:19:12] VERBOSE[4220] config.c: [Jan 13 14:19:12] == Found ... [Jan 13 14:19:12] VERBOSE[4220] pbx.c: [Jan 13 14:19:12] -- Including context 'TrunkAccounts' in context 'PROVIDERin' ... [Jan 13 14:19:12] VERBOSE[4220] pbx.c: [Jan 13 14:19:12] -- Registered extension context 'CheckOnNet' (0x2aaacc40d260) in local table 0xd0ba610; registrar: pbx_config [Jan 13 14:19:12] VERBOSE[4220] pbx.c: [Jan 13 14:19:12] -- Including context 'TrunkAccounts' in context 'CheckOnNet' ... Nothing special here it seems... Everything works fine when including context 'TrunkAccounts' in context 'PROVIDERin'. Here it functions as expected. But it does not work the same in context 'CheckOnNet'. Extra question : is there a difference between the context PROVIDERin and the procedure (sub) CheckOnNet ?? Kind regards, Jonas. -- _ -- 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 problem : not including context
Jonas Kellens wrote: Everything works fine when including context 'TrunkAccounts' in context 'PROVIDERin dialplan showPROVIDERin Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 problem : not including context
On 01/13/2012 02:59 PM, Doug Lytle wrote: Jonas Kellens wrote: Everything works fine when including context 'TrunkAccounts' in context 'PROVIDERin dialplan showPROVIDERin Doug Meaning ? -- _ -- 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 problem : not including context
Jonas Kellens wrote: Meaning ? Meaning I want to see the dialplan order of that context. I'm guessing that's your inbound context. With includes that also include sub-contexts. Usually, there is something ordered differently then expected. Also, what version of Asterisk? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 problem : not including context
On 01/13/2012 03:07 PM, Doug Lytle wrote:
Jonas Kellens wrote:
Meaning ?
Meaning I want to see the dialplan order of that context. I'm
guessing that's your inbound context. With includes that also include
sub-contexts.
Usually, there is something ordered differently then expected.
Also, what version of Asterisk?
Doug
Asterisk 1.6.2.22
It is impossible to post all of this information... However, this is the
context CheckOnNet :
...snip...
'_32962' = 1. GoTo(solutions,${EXTEN},1) [pbx_config]
'_3295[0-9]' = 1. GoTo(step,${EXTEN},1) [pbx_config]
'_.' = 1. NoOp()
[pbx_config]
2. Return()
[pbx_config]
Include ='TrunkAccounts'
[pbx_config]
-= 252 extensions (253 priorities) in 1 context. =-
Does this mean the Return() comes before Asterisk looks into the context
[TrunkAccounts] ??
Kind regards,
Jonas.
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Re: [asterisk-users] dialplan problem : not including context
Jonas Kellens wrote: Does this mean the Return() comes before Asterisk looks into the context [TrunkAccounts] ?? No, I believe the includes are read first, but the order in important. Since you may be matching against another context that may cause failure. For example, I have the following in a internal context: Include ='analog-extensions' [pbx_config] Include ='sip-utilities' [pbx_config] Include ='internal-extensions' [pbx_config] Include ='dial-local' [pbx_config] Include ='dial-ld' [pbx_config] Include ='incoming' [pbx_config] Include ='fall-through' [pbx_config] If I had fall-through first, it'd cause lots of issues. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 problem : not including context
On 01/13/2012 04:22 PM, Doug Lytle wrote: Jonas Kellens wrote: Does this mean the Return() comes before Asterisk looks into the context [TrunkAccounts] ?? No, I believe the includes are read first, but the order in important. Since you may be matching against another context that may cause failure. For example, I have the following in a internal context: Include ='analog-extensions' [pbx_config] Include ='sip-utilities' [pbx_config] Include ='internal-extensions' [pbx_config] Include ='dial-local' [pbx_config] Include ='dial-ld' [pbx_config] Include ='incoming' [pbx_config] Include ='fall-through' [pbx_config] If I had fall-through first, it'd cause lots of issues. Doug Hello, there is only one include-statement... Jonas. -- _ -- 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 problem : not including context
Jonas Kellens wrote: there is only one include-statement Then I don't know. I am still on 1.4.x and my PRI context contains all that I'm matching against (No sub contexts). Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 problem : not including context
Doug Lytle wrote: Then I don't know. I am still on 1.4.x and my PRI context contains all that I'm matching against (No sub contexts). One thing does come to mind; the inbound call is coming into your s extension and then your doing a gosub, in which case, you might be matching against s. Put in a few NoOP statements to find out what EXTEN or ARG1 is. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 problem : not including context
Le 13/01/2012 14:32, Jonas Kellens a écrit :
On 01/13/2012 02:23 PM, Doug Lytle wrote:
Jonas Kellens wrote:
I have the following in dialplan :
[TrunkAccounts]
dialplan show TrunkAccounts
Make sure the sort order is what you're expecting.
Doug
Hello,
The order is correct for as far as I'm sure.
[TrunkAccounts]
exten = 32380837,1,GoTo(01,32380837,1)
exten = 32380838,1,GoTo(01,32380838,1)
exten = 32380839,1,GoTo(01,32380839,1)
[CheckOnNet]
include = TrunkAccounts
exten = _321[0-3],1,GoTo(context1,${EXTEN},1)
exten = 3214,1,GoTo(context2, ${EXTEN} ,1)
exten = _.,1,NoOp()
exten = _.,n,Return()
Are you sure about your _. exten? Typo in the mail? It means 9
and more digits but your extensions are 8 digits ...
Include are always treated *after* context command. If _. is
right, something is wrong with Asterisk as it should treat
TrunkAccounts. If _XXX. (8 digits or more) is what you have in
yourdialplan, than the behavior of Asterisk is OK
Try
[TrunkAccounts]
exten = 32380837,1,GoTo(01,32380837,1)
exten = 32380838,1,GoTo(01,32380838,1)
exten = 32380839,1,GoTo(01,32380839,1)
[TrunkNotTreated]
exten = _.,1,NoOp()
exten = _.,n,Return()
[CheckOnNet]
include = TrunkAccounts
include = TrunkNotTreated
exten = _321[0-3],1,GoTo(context1,${EXTEN},1)
exten = 3214,1,GoTo(context2, ${EXTEN} ,1)
[...]
--
Daniel
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[asterisk-users] dialplan - dial command - custom ringtone
i could add r option in dial command. this will generate a ringtone during connection. could i change this default ringtone? i tried indications.conf but not success. -- _ -- 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 - dial command - custom ringtone
Hello Do you use hard phone or softphone? In many ip phones you can change the ring tones or use w option in Dial command Regards On Jan 3, 2012 4:08 AM, Qqblog Qqblog qqb...@ymail.com wrote: i could add r option in dial command. this will generate a ringtone during connection. could i change this default ringtone? i tried indications.conf but not success. -- _ -- 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 -- _ -- 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
[asterisk-users] Dialplan required for recording
Hi team, Can any one help me to implement dialplan in which conversation between a-party and b-party (call patch) needs to be recorded. Thanks Vinod Dharashive Sent from BlackBerry® on Airtel -- _ -- 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 required for recording
Hi Vinod, You Need to look in MIxmonitor application on asterisk. http://www.voip-info.org/wiki/view/MixMonitor http://www.the-asterisk-book.com/unstable/applikationen-mixmonitor.html Where you can find easy dialplan On Fri, Jul 29, 2011 at 4:35 AM, Vinod Dharashive vdharash...@gmail.comwrote: Hi team, Can any one help me to implement dialplan in which conversation between a-party and b-party (call patch) needs to be recorded. Thanks Vinod Dharashive Sent from BlackBerry® on Airtel -- _ -- 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 -- _ -- 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 pattern help
Thanks for the suggestion.
If I have to do this way i will check the AGI side...
Thanks
Armand
-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Leif Madsen
Envoyé : samedi 23 juillet 2011 20:18
À : asterisk-users@lists.digium.com
Objet : Re: [asterisk-users] dialplan pattern help
On 11-07-23 10:30 AM, Armand Fumal wrote:
Hi all,
I need help for make a pattern for a special case that i can't find the
solution.
In my case I want to match these in one pattern:
This is the same ext that can come in 4 cases
exten = _42704701,1,Macro(dialfax,${EXTEN:-8}) ; case with
42704701
exten = _X42704701,1,Macro(dialfax,${EXTEN:-8}) ; case with
042704701
exten = _42704701,1,Macro(dialfax,${EXTEN:-8}) ; case with +3242704701
exten = _XXX42704701,1,Macro(dialfax,${EXTEN:-8}); case with
3242704701
I have try _.42704701 but the parser stop to check after the point .:-(
So did you have any suggestion ?
Ya you can't match anything after the '.' in pattern matching.
I'm not sure the pattern matcher is really capable of doing what you want here.
The only way to do it really is to match less restrictively and perform a check
using dialplan applications/functions, and then if nothing is found, to fall
through.
Perhaps something like:
exten = _XXX,1,NoOp()
same = n,ExecIf($[${EXTEN:-8} = 42704701]?Macro(dialfax,${EXTEN:-8}))
same = n,Verbose(2,Did not match -- falling through)
same = n,Playback(invalid)
same = n,Hangup()
I'm pretty sure that's the only way you can do it in a single line (the
ExecIf() application).
Leif Madsen.
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[asterisk-users] dialplan pattern help
Hi all,
I need help for make a pattern for a special case that i can't find the
solution.
In my case I want to match these in one pattern:
This is the same ext that can come in 4 cases
exten = _42704701,1,Macro(dialfax,${EXTEN:-8}) ; case with 42704701
exten = _X42704701,1,Macro(dialfax,${EXTEN:-8}); case with
042704701
exten = _42704701,1,Macro(dialfax,${EXTEN:-8}) ; case with +3242704701
exten = _XXX42704701,1,Macro(dialfax,${EXTEN:-8}) ; case with
3242704701
I have try _.42704701 but the parser stop to check after the point .:-(
So did you have any suggestion ?
Regards
Armand Fumal
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Re: [asterisk-users] dialplan pattern help
On 11-07-23 10:30 AM, Armand Fumal wrote:
Hi all,
I need help for make a pattern for a special case that i can't find the
solution.
In my case I want to match these in one pattern:
This is the same ext that can come in 4 cases
exten = _42704701,1,Macro(dialfax,${EXTEN:-8}) ; case with
42704701
exten = _X42704701,1,Macro(dialfax,${EXTEN:-8}) ; case with
042704701
exten = _42704701,1,Macro(dialfax,${EXTEN:-8}) ; case with +3242704701
exten = _XXX42704701,1,Macro(dialfax,${EXTEN:-8}); case with
3242704701
I have try _.42704701 but the parser stop to check after the point .:-(
So did you have any suggestion ?
Ya you can't match anything after the '.' in pattern matching.
I'm not sure the pattern matcher is really capable of doing what you want here.
The only way to do it really is to match less restrictively and perform a check
using dialplan applications/functions, and then if nothing is found, to fall
through.
Perhaps something like:
exten = _XXX,1,NoOp()
same = n,ExecIf($[${EXTEN:-8} = 42704701]?Macro(dialfax,${EXTEN:-8}))
same = n,Verbose(2,Did not match -- falling through)
same = n,Playback(invalid)
same = n,Hangup()
I'm pretty sure that's the only way you can do it in a single line (the ExecIf()
application).
Leif Madsen.
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[asterisk-users] dialplan: all extern, except
Hi all, Perhaps a no-brainer, but i think i am making my dialplan on my proxy too complicated. Normally, what you find in the examples is that you have to dial a specific number, other 9 or 0 for an external line. What i want to do is this: If you pre-pend a number with something like * then you can dial local defined numbers, otherwise everything goes through my iax-trunk-line. So for instance: *#1 gives you a local welcome text *#2 gives you the local echo function while #1 gives you a remote welcome text #2 gives you the remote echo function And ordinary numbers or sip's go straight extern: 0174539053 or j.witvl...@a-domani.nl should go to my main asterisk-server. Currently i'm doing it pattern-matching all numbers, and each upper +lower case character, but i wonder if it can be done simpler. Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dialplan execution stops after ReceiveFax
Hello,
I have a noticed strange behavior in Asterisk 1.6.18.2 with ReceiveFax
Digium FAX Driver: 1.6.2.0_1.3.0 (optimized for i686_32).
I use a context [capi-in] for icoming ISDN calls:
==
[capi-in]
; Faxe fuer Ruben
exten = 12345,1,Macro(faxin,ruben.roeg...@jumping-frog.org,${EXTEN})
==
My macro for the fax receiving looks like that:
==
[macro-faxin]
; Faxe
; ARG1 = eMail-Adresse
exten = s,1,Verbose(${BOUNDARY} Eingehender Ruf von ${CALLERID(num)})
exten = s,n,Verbose(${BOUNDARY} BCHANNELINFO ${BCHANNELINFO})
; nur verarbeiten, wenn B-Kanal frei ist
exten = s,n,GotoIf($[${BCHANNELINFO} = 2]?hangup:free)
exten = s,n(free),NoOp()
exten = s,n,Set(TO=${ARG1})
exten = s,n,Verbose(1,${BOUNDARY} Eingehendes Fax ${CDR(uniqueid)})
exten = s,n,Set(FAXFILE=/var/spool/fax/fax-${TO}-${CDR(uniqueid)}.tif)
exten = s,n,Set(LOCALSTATIONID=jumping frog)
exten = s,n,Answer()
exten = s,n,Wait(3)
exten = s,n,ReceiveFAX(${FAXFILE},d)
exten = s,n,Verbose(1,${BOUNDARY} Nach dem Fax!)
exten = s,n,System(/usr/local/bin/fax2mail.sh ${FAXFILE} ${TO})
;exten = s,n,capicommand(receivefax,${FAXFILE},+00497613821,Headline,k)
exten = s,n(hangup),HangUp()
exten = h,1,System(/usr/local/bin/fax2mail.sh ${FAXFILE} ${TO})
==
As you can see, the received fax file should be processed by a
bash-script, but after the call hangs up, the script is never executed.
The console log shows:
==
-- Channel 'CAPI/ISDN1#02/3821-5' FAX session '4' is complete,
result: 'SUCCESS' (FAX_SUCCESS), error: 'NO_ERROR', pages: 1,
resolution: '204x98', transfer rate: '9600', remoteSID: '4932123847885'
== Spawn extension (macro-faxin, s, 11) exited non-zero on
'CAPI/ISDN1#02/12345-5' in macro 'faxin'
== Spawn extension (capi-in, 12345, 1) exited non-zero on
'CAPI/ISDN1#02/12345-5'
== ISDN1#02: CAPI Hangingup for PLCI=0x101 in state 2
ISDN1#02: CAPI INFO 0x3490: Normal call clearing
==
Anyone seeing what I'm missing?
Thank you.
Regards,
Ruben
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Re: [asterisk-users] dialplan execution stops after ReceiveFax
On 29/06/2011 5:13 PM, Ruben Rögels wrote:
Hello,
I have a noticed strange behavior in Asterisk 1.6.18.2 with ReceiveFax
Digium FAX Driver: 1.6.2.0_1.3.0 (optimized for i686_32).
I use a context [capi-in] for icoming ISDN calls:
==
[capi-in]
; Faxe fuer Ruben
exten = 12345,1,Macro(faxin,ruben.roeg...@jumping-frog.org,${EXTEN})
==
My macro for the fax receiving looks like that:
==
[macro-faxin]
; Faxe
; ARG1 = eMail-Adresse
exten = s,1,Verbose(${BOUNDARY} Eingehender Ruf von ${CALLERID(num)})
exten = s,n,Verbose(${BOUNDARY} BCHANNELINFO ${BCHANNELINFO})
; nur verarbeiten, wenn B-Kanal frei ist
exten = s,n,GotoIf($[${BCHANNELINFO} = 2]?hangup:free)
exten = s,n(free),NoOp()
exten = s,n,Set(TO=${ARG1})
exten = s,n,Verbose(1,${BOUNDARY} Eingehendes Fax ${CDR(uniqueid)})
exten = s,n,Set(FAXFILE=/var/spool/fax/fax-${TO}-${CDR(uniqueid)}.tif)
exten = s,n,Set(LOCALSTATIONID=jumping frog)
exten = s,n,Answer()
exten = s,n,Wait(3)
exten = s,n,ReceiveFAX(${FAXFILE},d)
exten = s,n,Verbose(1,${BOUNDARY} Nach dem Fax!)
exten = s,n,System(/usr/local/bin/fax2mail.sh ${FAXFILE} ${TO})
;exten = s,n,capicommand(receivefax,${FAXFILE},+00497613821,Headline,k)
exten = s,n(hangup),HangUp()
exten = h,1,System(/usr/local/bin/fax2mail.sh ${FAXFILE} ${TO})
==
As you can see, the received fax file should be processed by a
bash-script, but after the call hangs up, the script is never executed.
The console log shows:
==
-- Channel 'CAPI/ISDN1#02/3821-5' FAX session '4' is complete,
result: 'SUCCESS' (FAX_SUCCESS), error: 'NO_ERROR', pages: 1,
resolution: '204x98', transfer rate: '9600', remoteSID: '4932123847885'
== Spawn extension (macro-faxin, s, 11) exited non-zero on
'CAPI/ISDN1#02/12345-5' in macro 'faxin'
== Spawn extension (capi-in, 12345, 1) exited non-zero on
'CAPI/ISDN1#02/12345-5'
== ISDN1#02: CAPI Hangingup for PLCI=0x101 in state 2
ISDN1#02: CAPI INFO 0x3490: Normal call clearing
==
Anyone seeing what I'm missing?
Hi Ruben,
You should be looking at this thread
http://lists.digium.com/pipermail/asterisk-users/2011-June/263995.html
Presently I don't have the time to generate and send logs however soon
after my last post I did perform additional testing.
I am using ReceiveFAX using SPANDSP technology.
The occasions the System() call would not be executed, whether it was in
'h' of the dialplan or the main part of the macro after ReceiveFAX(),
was when a T.38 fax was being received, when it was a G.711 fax no
matter what I did to the call it would always execute the System() call
whether it was in the macro or the 'h'.
Cheers,
Larry.
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Re: [asterisk-users] dialplan execution stops after ReceiveFax
Hi Ruben,
You should be looking at this thread
http://lists.digium.com/pipermail/asterisk-users/2011-June/263995.html
Presently I don't have the time to generate and send logs however soon
after my last post I did perform additional testing.
I am using ReceiveFAX using SPANDSP technology.
The occasions the System() call would not be executed, whether it was in
'h' of the dialplan or the main part of the macro after ReceiveFAX(),
was when a T.38 fax was being received, when it was a G.711 fax no
matter what I did to the call it would always execute the System() call
whether it was in the macro or the 'h'.
Cheers,
Larry.
Hi Larry,
Ok. It looks like I don't understand the asterisks context and macro
system deep enough.
I tried to set
==
; Faxe fuer Ruben
exten = 3821,1,Macro(faxin,ruben.roeg...@jumping-frog.org,${EXTEN})
exten = h,1,System(/usr/local/bin/fax2mail.sh ${FAXFILE} ${TO})
==
Of couse I have to set up some number filtering but now, my bash script
is beeing executed.
So, as I understand the thread you gave me to read, the h-extension is
called in the context from which the macro is called and not in the
macro itself? This could explain why it works now.
Anyway, thank you for giving me the opportunity to look beyond my own
backyard!
Regards,
Ruben
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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
[asterisk-users] Dialplan matching
Hello all, I am trying to figure out the logic in on prefix matching for Asterisk 1.4.5. I want to be able to pass all international calls EXCEPT calls to 011870, 01137455 and so on. exten = _011870.,1,Goto(intl-disabled,s,1) exten = _01137455.,2,Goto(intl-disabled,s,1) exten = _01137477.,3,Goto(intl-disabled,s,1) exten = _0113749.,4,Goto(intl-disabled,s,1) exten = _011.,5,Goto(intl-disabled,s,1) exten = _011.,6,Playback(all-outgoing-lines-unavailable) exten = _011.,7,Wait(1) exten = _011.,8,Playback(please-hang-up-and-dial-operator) exten = _011.,9,Hangup Is this correct or should it be: exten = _011870X,1,Goto(intl-disabled,s,1) exten = _01137455X,2,Goto(intl-disabled,s,1) I tried searching for definitive information on voip-wiki, nerd vittles, but there is a lot of confusion. -- _ -- 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 matching
On Mon, Apr 4, 2011 at 8:09 AM, Asterisk User asteruserl...@gmail.comwrote: Hello all, I am trying to figure out the logic in on prefix matching for Asterisk 1.4.5. I want to be able to pass all international calls EXCEPT calls to 011870, 01137455 and so on. exten = _011870.,1,Goto(intl-disabled,s,1) exten = _01137455.,2,Goto(intl-disabled,s,1) exten = _01137477.,3,Goto(intl-disabled,s,1) exten = _0113749.,4,Goto(intl-disabled,s,1) exten = _011.,5,Goto(intl-disabled,s,1) exten = _011.,6,Playback(all-outgoing-lines-unavailable) exten = _011.,7,Wait(1) exten = _011.,8,Playback(please-hang-up-and-dial-operator) exten = _011.,9,Hangup Is this correct or should it be: exten = _011870X,1,Goto(intl-disabled,s,1) exten = _01137455X,2,Goto(intl-disabled,s,1) I tried searching for definitive information on voip-wiki, nerd vittles, but there is a lot of confusion. Assuming that 011870 is followed by more than digit, normally, I'd say your first set is more applicable. The . in the pattern at the end means any number of digits, followed by a timeout. If you know the number of digits, and it is fixed, then you could use _011870XXX or similar to avoid the timeout, and begin the Goto immediately on reception of the final digit. The X in the second set will match just one digit, and the Goto will be be executed. Does that help? -- 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 matching
On Monday 04 Apr 2011, Asterisk User wrote: Hello all, I am trying to figure out the logic in on prefix matching for Asterisk 1.4.5. I want to be able to pass all international calls EXCEPT calls to 011870, 01137455 and so on. Asterisk's default behaviour is always to try the hardest-to-match expression first (i.e. the exact extension number). If there is no match there, it then tries progressively easier-to-match expressions; only ever trying something like _. if nothing else matched. (Compare the rules of poker when wild cards are introduced: a natural hand beats an otherwise-equivalent hand containing wild cards.) -- AJS Answers come *after* questions. -- _ -- 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 matching
On Monday 04 April 2011 09:09:28 Asterisk User wrote: Hello all, I am trying to figure out the logic in on prefix matching for Asterisk 1.4.5. I want to be able to pass all international calls EXCEPT calls to 011870, 01137455 and so on. exten = _011870.,1,Goto(intl-disabled,s,1) This one is okay. exten = _01137455.,2,Goto(intl-disabled,s,1) Change this to priority 1. exten = _01137477.,3,Goto(intl-disabled,s,1) Change this to priority 1. exten = _0113749.,4,Goto(intl-disabled,s,1) Change this to priority 1. exten = _011.,5,Goto(intl-disabled,s,1) Change this to priority 1. exten = _011.,6,Playback(all-outgoing-lines-unavailable) exten = _011.,7,Wait(1) exten = _011.,8,Playback(please-hang-up-and-dial-operator) exten = _011.,9,Hangup This looks like it should be starting from priority 1, extension s, context [intl-disabled]. Is this correct or should it be: exten = _011870X,1,Goto(intl-disabled,s,1) exten = _01137455X,2,Goto(intl-disabled,s,1) I tried searching for definitive information on voip-wiki, nerd vittles, but there is a lot of confusion. The major problem in your dialplan is that you WANT to have multiple start points, but the way you have it written, there is only ONE start point. Everything else is simply ignored. Extensions will only start in the dialplan from priority 1. -- Tilghman -- _ -- 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
