[asterisk-users] AEL2: How to get rid of "does not end with a return; I will insert one" warnings
Hi,
I'm converting to asterisk 1.8 an existing (and lengthy) dialplan
written in AEL2.
I'm using in many places things like
macro foo {
BlahBlah();
return;
catch h {
BlahBlah();
}
};
How can I safely get rid of "does not end with a return; I will insert
one" warnings with such constructs ?
Regards
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[asterisk-users] AEL2: How get rid of "expression has operators, but no variables" warnings
Hi, I'm converting an existing dialplan written in AEL2 to Asterisk 1.8. While at it, I'm trying to get rid of some AEL2 warnings messages that used to clutter my console when loading AEL scripts. Specifically, my dialplan includes this simple text assignment : prefix1="FooBar => "; It produces this: expression "FooBar => " has operators, but no variables. Interesting... Using Set application makes it but I would appreciate an other way to do it. Suggestion ? Regards -- _ -- 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] AEL2, BASE64_DECODE and hexadecimal
On Thursday 09 April 2009 10:14:12 Tilghman Lesher wrote: > On Thursday 09 April 2009 04:28:16 Olivier wrote: > > 2009/4/8 Tilghman Lesher > > > > > On Tuesday 07 April 2009 23:38:08 Olivier wrote: > > > > 2009/4/7 Mark Michelson > > > > > > > > > Philipp Kempgen wrote: > > > > > > BTW (developer's question) is there a reason why SendText() resp. > > > > > > sendtext_exec() refuses to send zero-length data? > > > > > > > > > > I can't point to any specific reason. I assume that whoever wrote > > > > > the application probably thought that attempting to send > > > > > zero-length data > > > > > > was > > > > > > > > pointless and that if no data were passed to the application, it > > > > > likely was due > > > > > to an error by the user. > > > > > > > > The phone I'm working on (Thomson ST2030) would display in slow > > > > blinking, inversed letters (white on black) any text received in SIP > > > > MESSAGE. Display duration is unlimited. > > > > To erase an old message, you must send a single carriage return (or > > > > maybe an empty string). > > > > > > > > I'm wondering how many phones behave like this ? > > > > > > > > Maybe, sendtext should then be refactored to accommodate this. > > > > > > What does the phone do when you send a single space? > > > > It would display it as a black rectangle : as unfortunately, this phone > > displays text in inversed color (white letters on black background), a > > single space remains visible. > > > > It seems I really need to send a carriage return ($0D in hexa) but I > > couldn't find a way to pass such string using SendText. > > > > As suggested earlier, maybe an empty SIP MESSAGE would do the trick but > > as SendText rejects empty strings, I've not tested it yet. For that, > > maybe I could try to build my own custom SIP MESSAGE, faking a true SIP > > MESSAGE to double check if an empty string would be acceptable solution ? > > > > (Note, that sending a carriage return must work as vendor handed to me a > > Wireshark capture for a "working SIP MESSAGE") > > I suspect not. You really need to send an empty message; that is, a > message with a content-length of 0. A message containing a carriage return > would still have a content-length of 1 and would confuse the phone at > worst, and display an unprintable character, at best. > > You could send an empty message in SendText with SendText(,), but chan_sip > still doesn't allow the empty message, so a code change is still necessary. Fixed in 1.4 in changeset 187362. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL2, BASE64_DECODE and hexadecimal
On Thursday 09 April 2009 04:28:16 Olivier wrote: > 2009/4/8 Tilghman Lesher > > > On Tuesday 07 April 2009 23:38:08 Olivier wrote: > > > 2009/4/7 Mark Michelson > > > > > > > Philipp Kempgen wrote: > > > > > BTW (developer's question) is there a reason why SendText() resp. > > > > > sendtext_exec() refuses to send zero-length data? > > > > > > > > I can't point to any specific reason. I assume that whoever wrote the > > > > application probably thought that attempting to send zero-length data > > > > was > > > > > > pointless and that if no data were passed to the application, it > > > > likely was due > > > > to an error by the user. > > > > > > The phone I'm working on (Thomson ST2030) would display in slow > > > blinking, inversed letters (white on black) any text received in SIP > > > MESSAGE. Display duration is unlimited. > > > To erase an old message, you must send a single carriage return (or > > > maybe an empty string). > > > > > > I'm wondering how many phones behave like this ? > > > > > > Maybe, sendtext should then be refactored to accommodate this. > > > > What does the phone do when you send a single space? > > It would display it as a black rectangle : as unfortunately, this phone > displays text in inversed color (white letters on black background), a > single space remains visible. > > It seems I really need to send a carriage return ($0D in hexa) but I > couldn't find a way to pass such string using SendText. > > As suggested earlier, maybe an empty SIP MESSAGE would do the trick but as > SendText rejects empty strings, I've not tested it yet. For that, maybe I > could try to build my own custom SIP MESSAGE, faking a true SIP MESSAGE to > double check if an empty string would be acceptable solution ? > > (Note, that sending a carriage return must work as vendor handed to me a > Wireshark capture for a "working SIP MESSAGE") I suspect not. You really need to send an empty message; that is, a message with a content-length of 0. A message containing a carriage return would still have a content-length of 1 and would confuse the phone at worst, and display an unprintable character, at best. You could send an empty message in SendText with SendText(,), but chan_sip still doesn't allow the empty message, so a code change is still necessary. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL2, BASE64_DECODE and hexadecimal
2009/4/8 Tilghman Lesher > On Tuesday 07 April 2009 23:38:08 Olivier wrote: > > 2009/4/7 Mark Michelson > > > > > Philipp Kempgen wrote: > > > > BTW (developer's question) is there a reason why SendText() resp. > > > > sendtext_exec() refuses to send zero-length data? > > > > > > I can't point to any specific reason. I assume that whoever wrote the > > > application probably thought that attempting to send zero-length data > was > > > pointless and that if no data were passed to the application, it likely > > > was due > > > to an error by the user. > > > > The phone I'm working on (Thomson ST2030) would display in slow blinking, > > inversed letters (white on black) any text received in SIP MESSAGE. > > Display duration is unlimited. > > To erase an old message, you must send a single carriage return (or maybe > > an empty string). > > > > I'm wondering how many phones behave like this ? > > > > Maybe, sendtext should then be refactored to accommodate this. > > What does the phone do when you send a single space? It would display it as a black rectangle : as unfortunately, this phone displays text in inversed color (white letters on black background), a single space remains visible. It seems I really need to send a carriage return ($0D in hexa) but I couldn't find a way to pass such string using SendText. As suggested earlier, maybe an empty SIP MESSAGE would do the trick but as SendText rejects empty strings, I've not tested it yet. For that, maybe I could try to build my own custom SIP MESSAGE, faking a true SIP MESSAGE to double check if an empty string would be acceptable solution ? (Note, that sending a carriage return must work as vendor handed to me a Wireshark capture for a "working SIP MESSAGE") > > -- > Tilghman > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL2, BASE64_DECODE and hexadecimal
On Tuesday 07 April 2009 23:38:08 Olivier wrote: > 2009/4/7 Mark Michelson > > > Philipp Kempgen wrote: > > > BTW (developer's question) is there a reason why SendText() resp. > > > sendtext_exec() refuses to send zero-length data? > > > > I can't point to any specific reason. I assume that whoever wrote the > > application probably thought that attempting to send zero-length data was > > pointless and that if no data were passed to the application, it likely > > was due > > to an error by the user. > > The phone I'm working on (Thomson ST2030) would display in slow blinking, > inversed letters (white on black) any text received in SIP MESSAGE. > Display duration is unlimited. > To erase an old message, you must send a single carriage return (or maybe > an empty string). > > I'm wondering how many phones behave like this ? > > Maybe, sendtext should then be refactored to accommodate this. What does the phone do when you send a single space? -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL2, BASE64_DECODE and hexadecimal
2009/4/7 Mark Michelson
> Philipp Kempgen wrote:
> > Olivier schrieb:
> >> 2009/4/7 Philipp Kempgen
> >>> Olivier schrieb:
> I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext
> in
> >>> an
> AEL2 file like this :
> SendText(${BASE64_DECODE(DQ==)});
>
>
> Value sent (8 bytes long) is very strange :
> Content-Type: text/plain;charset=UTF-8
> Content-Length: 8
>
> �ez?==
> >>> I doubt you will find a way to properly pass CR or LF to an
> >>> application in extensions.(conf|ael) but keep us in the loop.
> >
> >> It's strange how such a silly thing is somehow keeping me from centrally
> >> managing phones forwarding : I can display a phone is forwarded but I
> can't
> >> gracefully return to previous status ...
> >
> > BTW (developer's question) is there a reason why SendText() resp.
> > sendtext_exec() refuses to send zero-length data?
> >
>
> I can't point to any specific reason. I assume that whoever wrote the
> application probably thought that attempting to send zero-length data was
> pointless and that if no data were passed to the application, it likely was
> due
> to an error by the user.
The phone I'm working on (Thomson ST2030) would display in slow blinking,
inversed letters (white on black) any text received in SIP MESSAGE.
Display duration is unlimited.
To erase an old message, you must send a single carriage return (or maybe an
empty string).
I'm wondering how many phones behave like this ?
Maybe, sendtext should then be refactored to accommodate this.
>
>
> Mark Michelson
>
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Re: [asterisk-users] AEL2, BASE64_DECODE and hexadecimal
Philipp Kempgen wrote:
> Olivier schrieb:
>> 2009/4/7 Philipp Kempgen
>>> Olivier schrieb:
I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in
>>> an
AEL2 file like this :
SendText(${BASE64_DECODE(DQ==)});
Value sent (8 bytes long) is very strange :
Content-Type: text/plain;charset=UTF-8
Content-Length: 8
�ez?==
>>> I doubt you will find a way to properly pass CR or LF to an
>>> application in extensions.(conf|ael) but keep us in the loop.
>
>> It's strange how such a silly thing is somehow keeping me from centrally
>> managing phones forwarding : I can display a phone is forwarded but I can't
>> gracefully return to previous status ...
>
> BTW (developer's question) is there a reason why SendText() resp.
> sendtext_exec() refuses to send zero-length data?
>
I can't point to any specific reason. I assume that whoever wrote the
application probably thought that attempting to send zero-length data was
pointless and that if no data were passed to the application, it likely was due
to an error by the user.
Mark Michelson
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Re: [asterisk-users] AEL2, BASE64_DECODE and hexadecimal
Olivier schrieb:
> 2009/4/7 Philipp Kempgen
>> Olivier schrieb:
>> > I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in
>> an
>> > AEL2 file like this :
>> > SendText(${BASE64_DECODE(DQ==)});
>> >
>> >
>> > Value sent (8 bytes long) is very strange :
>> > Content-Type: text/plain;charset=UTF-8
>> > Content-Length: 8
>> >
>> > �ez?==
>>
>> I doubt you will find a way to properly pass CR or LF to an
>> application in extensions.(conf|ael) but keep us in the loop.
> It's strange how such a silly thing is somehow keeping me from centrally
> managing phones forwarding : I can display a phone is forwarded but I can't
> gracefully return to previous status ...
BTW (developer's question) is there a reason why SendText() resp.
sendtext_exec() refuses to send zero-length data?
Philipp Kempgen
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Re: [asterisk-users] AEL2, BASE64_DECODE and hexadecimal
2009/4/7 Philipp Kempgen
> Olivier schrieb:
> > I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in
> an
> > AEL2 file like this :
> > SendText(${BASE64_DECODE(DQ==)});
> >
> >
> > Value sent (8 bytes long) is very strange :
> > Content-Type: text/plain;charset=UTF-8
> > Content-Length: 8
> >
> > �ez?==
>
> I doubt you will find a way to properly pass CR or LF to an
> application in extensions.(conf|ael) but keep us in the loop.
I broke my teeth on that all day long !
I was about to try AGI ...
It's strange how such a silly thing is somehow keeping me from centrally
managing phones forwarding : I can display a phone is forwarded but I can't
gracefully return to previous status ...
>
>
>Philipp Kempgen
> --
> AMOOCON 2009, May 4-5, Rostock / Germany -> http://www.amoocon.de
> Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
> AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de
> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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>
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Re: [asterisk-users] AEL2, BASE64_DECODE and hexadecimal
Olivier schrieb:
> I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in an
> AEL2 file like this :
> SendText(${BASE64_DECODE(DQ==)});
>
>
> Value sent (8 bytes long) is very strange :
> Content-Type: text/plain;charset=UTF-8
> Content-Length: 8
>
> �ez?==
I doubt you will find a way to properly pass CR or LF to an
application in extensions.(conf|ael) but keep us in the loop.
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany -> http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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[asterisk-users] AEL2, BASE64_DECODE and hexadecimal
Hi,
I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in an
AEL2 file like this :
SendText(${BASE64_DECODE(DQ==)});
Value sent (8 bytes long) is very strange :
Content-Type: text/plain;charset=UTF-8
Content-Length: 8
�ez?==
Any workaround ?
Regards
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Re: [asterisk-users] AEL2: If-then-else not permitted in Switch-Case
Thanks guys.
It was the If vs if that was causing the problem.
This is probably due to my good coding practice of other languages in
the past :-)
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Watkins, Bradley
> Sent: Thursday, 5 March 2009 9:12 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] AEL2: If-then-else not permitted in
Switch-
> Case
>
> > I just want to confirm but it seems that if-then-else is not
permitted
> > in case structure.
> > It was not really documented but it seems to be the case.
> >
> > Can anyone confirm?
>
> No, if-then-else works fine inside a case statement. See inline
> comments.
> >
> > switch(${DIALSTATUS})
> > {
> > case NOANSWER:
> > {
> This brace, and its closing-brace mate, are superfluous though not
> harmful.
>
> >// if-then-else not permitted
> >If (${ael-var} = 1)
> Your primary problem is probably right here, the if needs to be all
> lower-case ( If != if ).
>
> >{
> > Playback(beep);
> > return;
> >}
> > }
> Again, unnecessary.
>
> > case BUSY:
> > {
> >return;
> > }
> > default:
> > {
> >Hangup();
> > };
> > }
> >
> >
> >
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Re: [asterisk-users] AEL2: If-then-else not permitted in Switch-Case
> I just want to confirm but it seems that if-then-else is not permitted
> in case structure.
> It was not really documented but it seems to be the case.
>
> Can anyone confirm?
No, if-then-else works fine inside a case statement. See inline
comments.
>
> switch(${DIALSTATUS})
> {
> case NOANSWER:
> {
This brace, and its closing-brace mate, are superfluous though not
harmful.
>// if-then-else not permitted
>If (${ael-var} = 1)
Your primary problem is probably right here, the if needs to be all
lower-case ( If != if ).
>{
> Playback(beep);
> return;
>}
> }
Again, unnecessary.
> case BUSY:
> {
>return;
> }
> default:
> {
>Hangup();
> };
> }
>
>
>
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>
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Re: [asterisk-users] AEL2: If-then-else not permitted in Switch-Case
Philipp Kempgen wrote: > Courier mail server at exa.billmerriam.com schrieb: > >> This is a delivery status notification from exa.billmerriam.com, >> running the Courier mail server, version 0.54.1. >> >> The original message was received on Wed, 04 Mar 2009 09:10:55 -0500 >> from localhost (localhost [127.0.0.1]) >> >> --- >> >>UNDELIVERABLE MAIL >> >> Your message to the following recipients cannot be delivered: >> >> : >> yocto.billmerriam.com [68.209.186.200]: >> STARTTLS >> <<< 500 couriertls: connect: Connection reset by peer >> > > li...@billmerriam.com, please fix your mail server. > I sent the message to the asterisk-users mailing list and - sorry > to say - I don't care if it was delivered to you or not. > > Thanks, > > Philipp Kempgen > ...and so you replied to it? I mean if he didn't get the original copy, he sure isn't going to get your terse reply. The rest of us however -- Thank you and have any kind of day you want, Anthony Francis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL2: If-then-else not permitted in Switch-Case
Courier mail server at exa.billmerriam.com schrieb: > This is a delivery status notification from exa.billmerriam.com, > running the Courier mail server, version 0.54.1. > > The original message was received on Wed, 04 Mar 2009 09:10:55 -0500 > from localhost (localhost [127.0.0.1]) > > --- > >UNDELIVERABLE MAIL > > Your message to the following recipients cannot be delivered: > > : > yocto.billmerriam.com [68.209.186.200]: >>> STARTTLS > <<< 500 couriertls: connect: Connection reset by peer li...@billmerriam.com, please fix your mail server. I sent the message to the asterisk-users mailing list and - sorry to say - I don't care if it was delivered to you or not. Thanks, Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany -> http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL2: If-then-else not permitted in Switch-Case
Klaus Darilion schrieb:
> Lee, John (Sydney) schrieb:
>> I just want to confirm but it seems that if-then-else is not permitted
>> in case structure.
>> It was not really documented but it seems to be the case.
>>
>> Can anyone confirm?
>>
>> switch(${DIALSTATUS})
>> {
>> case NOANSWER:
>> {
^ no code block required here.
probably invalid syntax.
>>// if-then-else not permitted
>>If (${ael-var} = 1)
>
>^^ case sensitive?
>
>>{
>> Playback(beep);
>> return;
>>}
>> }
>> case BUSY:
>> {
>>return;
>> }
>> default:
>> {
>>Hangup();
>> };
>> }
Try `aelparse -n`
Philipp Kempgen
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Re: [asterisk-users] AEL2: If-then-else not permitted in Switch-Case
Lee, John (Sydney) schrieb:
> I just want to confirm but it seems that if-then-else is not permitted
> in case structure.
> It was not really documented but it seems to be the case.
>
> Can anyone confirm?
>
> switch(${DIALSTATUS})
> {
> case NOANSWER:
> {
>// if-then-else not permitted
>If (${ael-var} = 1)
^^ case sensitive?
>{
> Playback(beep);
> return;
>}
> }
> case BUSY:
> {
>return;
> }
> default:
> {
>Hangup();
> };
> }
>
>
>
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Re: [asterisk-users] AEL2: If-then-else not permitted in Switch-Case
Lee, John (Sydney) schrieb:
> I just want to confirm but it seems that if-then-else is not permitted
> in case structure.
> It was not really documented but it seems to be the case.
>
> Can anyone confirm?
>
> switch(${DIALSTATUS})
> {
> case NOANSWER:
> {
>// if-then-else not permitted
>If (${ael-var} = 1)
>{
> Playback(beep);
> return;
>}
> }
I would have written this like so:
switch ("${DIALSTATUS}") {
case "NOANSWER":
if ("${ael-var}" = "1") {
Playback(beep);
}
break;
}
Give it a try.
Philipp Kempgen
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[asterisk-users] AEL2: If-then-else not permitted in Switch-Case
I just want to confirm but it seems that if-then-else is not permitted
in case structure.
It was not really documented but it seems to be the case.
Can anyone confirm?
switch(${DIALSTATUS})
{
case NOANSWER:
{
// if-then-else not permitted
If (${ael-var} = 1)
{
Playback(beep);
return;
}
}
case BUSY:
{
return;
}
default:
{
Hangup();
};
}
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[asterisk-users] AEL2 Hint & Parking
I've been reading most of the day and can't seem to find a clear
definition of the syntax for parking lot hints in AEL2. I have tried
all of the following and they either do not light up the line button on
my Snom 300 or give syntax errors:
hint(park/701) 701 => {
ParkedCall(701);
}
hint(park:701) 701 => {
ParkedCall(701);
}
hint(park/[EMAIL PROTECTED]) 701 => {
ParkedCall(701);
}
hint(park:[EMAIL PROTECTED]) 701 => {
ParkedCall(701);
}
I have this in my context as well:
includes {
parkedcalls;
}
I do not see any indication on the CLI that Asterisk is attempting to
notify my sip phone of the status change and I have verbose and debug at 20.
Any ideas?
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Re: [asterisk-users] AEL2 and Callbacks
Douglas Garstang wrote: > I am originating a command via the AMI with this... Doug, Were you ever able to resolve this? If so, could you share what the issue was? Thanks, Sean ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL2 and Callbacks
Do a 'core show dialplan' and see what the AEL is generating.
On 11/1/07, Douglas Garstang <[EMAIL PROTECTED]> wrote:
>
> >- Original Message
> >From: Richard Lyman <[EMAIL PROTECTED]>
> >To: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> >Sent: Thursday, November 1, 2007 8:47:28 AM
> >Subject: Re: [asterisk-users] AEL2 and Callbacks
> >
> >Douglas Garstang wrote:
> >> I am originating a command via the AMI with this...
> >>
> >> Action: Login
> >> Username: xxx
> >> Secret: yyy
> >>
> >> ACTION: Originate
> >> Async: yes
> >> Timeout: 6
> >> Exten: callback
> >> Channel: Local/[EMAIL PROTECTED]
> >> Callerid: 849120
> >> Context: default
> >> ActionID: 849120
> >>
> >> My LegA context:
> >> ---
> >> context LegA {
> >>_X. => {
> >>Dial(SIP/[EMAIL PROTECTED]);
> >>}
> >>
> >> }
> >>
> >> And my default context:
> >> --
> >> context default {
> >>callback => {
> >>NoCDR();
> >>Wait(1);
> >>
> Dial(${destination},60,oL(${timeout}:${timeout_warning}:${timeout_warning_repeat}));
> >>}
> >>
> >> }
> >>
> >> The A leg is established, and once Asterisk goes to dial the B leg...
> >>
> >>-- Executing [EMAIL PROTECTED]:1] Dial("Local/[EMAIL PROTECTED],2",
> "SIP/[EMAIL PROTECTED]") in new stack
> >>-- Called [EMAIL PROTECTED]
> >>-- SIP/Provider-09a8cff8 is making progress passing it to
> Local/[EMAIL PROTECTED],2
> >>
> > -- SIP/Provider-09a8cff8 answered Local/[EMAIL PROTECTED],2
> >> == Starting Local/[EMAIL PROTECTED],1 at default,callback,1 failed
> so falling back to exten 's'
> >> == Starting Local/[EMAIL PROTECTED],1 at default,s,1 still failed
> so falling back to context 'default'
> >> [Oct 31 01:57:07] WARNING[29795]: pbx.c:2450 __ast_pbx_run: Channel '
> Local/[EMAIL PROTECTED],1' sent into invalid extension 's' in context
> 'default', >but no invalid handler
> >>
> >> Uhm, why? I have a default context with a callback extension. Of course
> I have no explicit priority 1 though... this is AEL2
> >> What's it complaining for?
> >>
> >> Doug.
> >>
> >>
> >>
> >originates have always had an issue where it falls back to an 's'
> >extension. and since you do not have one, nor an 'i' for invalid
> >extension... it bombs out.
>
> Yes... but I DO have a default context and I DO have a callback extension.
> What's it whining about?
>
> Doug.
>
>
>
>
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Re: [asterisk-users] AEL2 and Callbacks
>- Original Message
>From: Richard Lyman <[EMAIL PROTECTED]>
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>
>Sent: Thursday, November 1, 2007 8:47:28 AM
>Subject: Re: [asterisk-users] AEL2 and Callbacks
>
>Douglas Garstang wrote:
>> I am originating a command via the AMI with this...
>>
>> Action: Login
>> Username: xxx
>> Secret: yyy
>>
>> ACTION: Originate
>> Async: yes
>> Timeout: 6
>> Exten: callback
>> Channel: Local/[EMAIL PROTECTED]
>> Callerid: 849120
>> Context: default
>> ActionID: 849120
>>
>> My LegA context:
>> ---
>> context LegA {
>> _X. => {
>> Dial(SIP/[EMAIL PROTECTED]);
>> }
>>
>> }
>>
>> And my default context:
>> --
>> context default {
>> callback => {
>> NoCDR();
>> Wait(1);
>>
Dial(${destination},60,oL(${timeout}:${timeout_warning}:${timeout_warning_repeat}));
>> }
>>
>> }
>>
>> The A leg is established, and once Asterisk goes to dial the B leg...
>>
>> -- Executing [EMAIL PROTECTED]:1]
Dial("Local/[EMAIL PROTECTED],2", "SIP/[EMAIL PROTECTED]") in new stack
>> -- Called [EMAIL PROTECTED]
>> -- SIP/Provider-09a8cff8 is making progress passing it to
Local/[EMAIL PROTECTED],2
>>
>-- SIP/Provider-09a8cff8 answered Local/[EMAIL PROTECTED],2
>> == Starting Local/[EMAIL PROTECTED],1 at default,callback,1
failed so falling back to exten 's'
>> == Starting Local/[EMAIL PROTECTED],1 at default,s,1 still
failed so falling back to context 'default'
>> [Oct 31 01:57:07] WARNING[29795]: pbx.c:2450 __ast_pbx_run: Channel
'Local/[EMAIL PROTECTED],1' sent into invalid extension 's' in
context 'default', >but no invalid handler
>>
>> Uhm, why? I have a default context with a callback extension. Of
course I have no explicit priority 1 though... this is AEL2
>> What's it complaining for?
>>
>> Doug.
>>
>>
>>
>originates have always had an issue where it falls back to an 's'
>extension. and since you do not have one, nor an 'i' for invalid
>extension... it bombs out.
Yes... but I DO have a default context and I DO have a callback extension.
What's it whining about?
Doug.
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Re: [asterisk-users] AEL2 and Callbacks
Douglas Garstang wrote:
> I am originating a command via the AMI with this...
>
> Action: Login
> Username: xxx
> Secret: yyy
>
> ACTION: Originate
> Async: yes
> Timeout: 6
> Exten: callback
> Channel: Local/[EMAIL PROTECTED]
> Callerid: 849120
> Context: default
> ActionID: 849120
>
> My LegA context:
> ---
> context LegA {
> _X. => {
> Dial(SIP/[EMAIL PROTECTED]);
> }
>
> }
>
> And my default context:
> --
> context default {
> callback => {
> NoCDR();
> Wait(1);
>
> Dial(${destination},60,oL(${timeout}:${timeout_warning}:${timeout_warning_repeat}));
> }
>
> }
>
> The A leg is established, and once Asterisk goes to dial the B leg...
>
> -- Executing [EMAIL PROTECTED]:1] Dial("Local/[EMAIL PROTECTED],2",
> "SIP/[EMAIL PROTECTED]") in new stack
> -- Called [EMAIL PROTECTED]
> -- SIP/Provider-09a8cff8 is making progress passing it to Local/[EMAIL
> PROTECTED],2
> -- SIP/Provider-09a8cff8 answered Local/[EMAIL PROTECTED],2
> == Starting Local/[EMAIL PROTECTED],1 at default,callback,1 failed so
> falling back to exten 's'
> == Starting Local/[EMAIL PROTECTED],1 at default,s,1 still failed so
> falling back to context 'default'
> [Oct 31 01:57:07] WARNING[29795]: pbx.c:2450 __ast_pbx_run: Channel
> 'Local/[EMAIL PROTECTED],1' sent into invalid extension 's' in context
> 'default', but no invalid handler
>
> Uhm, why? I have a default context with a callback extension. Of course I
> have no explicit priority 1 though... this is AEL2
> What's it complaining for?
>
> Doug.
>
>
>
originates have always had an issue where it falls back to an 's'
extension. and since you do not have one, nor an 'i' for invalid
extension... it bombs out.
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Re: [asterisk-users] AEL2 and Callbacks
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 What do you get if you do dialplan show default? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHKUlTDQNt8rg0Kp4RAteuAJ9kbLC77Bw7G789uOIaQ1hR+++87gCgqNPB p4jMkvOg6kuVylFKaHLPwAs= =ajMg -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL2 and Callbacks
I am originating a command via the AMI with this...
Action: Login
Username: xxx
Secret: yyy
ACTION: Originate
Async: yes
Timeout: 6
Exten: callback
Channel: Local/[EMAIL PROTECTED]
Callerid: 849120
Context: default
ActionID: 849120
My LegA context:
---
context LegA {
_X. => {
Dial(SIP/[EMAIL PROTECTED]);
}
}
And my default context:
--
context default {
callback => {
NoCDR();
Wait(1);
Dial(${destination},60,oL(${timeout}:${timeout_warning}:${timeout_warning_repeat}));
}
}
The A leg is established, and once Asterisk goes to dial the B leg...
-- Executing [EMAIL PROTECTED]:1] Dial("Local/[EMAIL PROTECTED],2",
"SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/Provider-09a8cff8 is making progress passing it to Local/[EMAIL
PROTECTED],2
-- SIP/Provider-09a8cff8 answered Local/[EMAIL PROTECTED],2
== Starting Local/[EMAIL PROTECTED],1 at default,callback,1 failed so falling
back to exten 's'
== Starting Local/[EMAIL PROTECTED],1 at default,s,1 still failed so falling
back to context 'default'
[Oct 31 01:57:07] WARNING[29795]: pbx.c:2450 __ast_pbx_run: Channel
'Local/[EMAIL PROTECTED],1' sent into invalid extension 's' in context
'default', but no invalid handler
Uhm, why? I have a default context with a callback extension. Of course I have
no explicit priority 1 though... this is AEL2
What's it complaining for?
Doug.
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Re: [asterisk-users] AEL2 Syntax Highlighting
On Mon, 2007-10-15 at 11:42 -0500, Perssy Llamosas wrote:
> Original Message
> Subject: Re:[asterisk-users] AEL2 Syntax Highlighting
> From: Tzafrir Cohen <[EMAIL PROTECTED]>
> To: asterisk-users@lists.digium.com
> Date: 13/10/2007 05:24 a.m.
> > On Fri, Oct 12, 2007 at 05:24:29PM -0500, Perssy Llamosas wrote:
> >
> >> Hi,
> >>
> >> I am looking for a syntax highlighter for AEL2. Google is not helping,
> >> so I thought you guys could help me.
> >>
> >> I found this vim syntax highlighter for AEL but it doesn't help if you
> >> want to code in AEL2:
> >> http://vim.sourceforge.net/scripts/script.php?script_id=1900
> >>
> >
> > How is AEL2 syntax different from AEL?
> >
> > Can you please give examples where the above fails for AEL2? (or for
> > AEL, for that matter)
> >
> Well, I am trying to improve that script slowly, I admit I knew nothing
> about writing vim highlighting files before so this is a good
> opportunity to learn...
>
> Some examples where the above fails:
> //-example: No ";" after brackets.
This is true; in the original AEL, a semicolon must follow EVERY
statement, even statement blocks. IN AEL2, the ';' after a statement
block is treated as an empty statement, and is ignored.
I am sure you will bump into other subtle differences... If I were to
try to enumerate them, I'd probably miss something, but I did give a
list of differences in the AEL2 voip-info Wiki... I congratulate you on
taking on this effort, and wish you all the success possible. If you
have any questions, feel free to write/irc/jabber/phone me. I'm
'usually' around.
murf
> globals {
> }
> /* Anything below fails */
> context failed1 {
> };
>
> //-example: No ";" after brackets.
> context failed2 {
> 1 => {
>Hangup();
> }
> /* Anything below fails */
> 2 => {
>Hangup();
> };
> };
>
> //-example: Inline if else while for random
> context failed3 {
> 1 => {
>if(1) NoOp(This fails);
> };
> 2 => {
>if(1) {NoOp(This also fails);}
> };
> };
>
> //-example: bug
> context failed4 {
> 1 => {
>if (1) {
>} else {
>}
>/* Anything below fails */
> };
> 2 => {
>NoOp(This fails);
> };
> };
>
> //-example: bug
> context failed5 {
> 1 => {
>switch(1) {
>}
>/* Anything below fails */
> };
> 2 => {
>NoOp(This fails);
> };
> };
>
> //-example: Hints
> context failed6 {
> hint(Sip/1) 2 => {
>NoOp(This fails);
> };
> };
>
> //-example: Next line bracket
> context failed7
> {
> 1 => {
>NoOp(This fails);
> };
> };
>
> //-example: Switches and eswitches
> context failed8 {
> switches {
> IAX2/abox;
> };
> /* Anything below fails */
> 1 => {
> };
> };
>
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Re: [asterisk-users] AEL2 Syntax Highlighting
Original Message
Subject: Re:[asterisk-users] AEL2 Syntax Highlighting
From: Tzafrir Cohen <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Date: 13/10/2007 05:24 a.m.
> On Fri, Oct 12, 2007 at 05:24:29PM -0500, Perssy Llamosas wrote:
>
>> Hi,
>>
>> I am looking for a syntax highlighter for AEL2. Google is not helping,
>> so I thought you guys could help me.
>>
>> I found this vim syntax highlighter for AEL but it doesn't help if you
>> want to code in AEL2:
>> http://vim.sourceforge.net/scripts/script.php?script_id=1900
>>
>
> How is AEL2 syntax different from AEL?
>
> Can you please give examples where the above fails for AEL2? (or for
> AEL, for that matter)
>
Well, I am trying to improve that script slowly, I admit I knew nothing
about writing vim highlighting files before so this is a good
opportunity to learn...
Some examples where the above fails:
//-example: No ";" after brackets.
globals {
}
/* Anything below fails */
context failed1 {
};
//-example: No ";" after brackets.
context failed2 {
1 => {
Hangup();
}
/* Anything below fails */
2 => {
Hangup();
};
};
//-example: Inline if else while for random
context failed3 {
1 => {
if(1) NoOp(This fails);
};
2 => {
if(1) {NoOp(This also fails);}
};
};
//-example: bug
context failed4 {
1 => {
if (1) {
} else {
}
/* Anything below fails */
};
2 => {
NoOp(This fails);
};
};
//-example: bug
context failed5 {
1 => {
switch(1) {
}
/* Anything below fails */
};
2 => {
NoOp(This fails);
};
};
//-example: Hints
context failed6 {
hint(Sip/1) 2 => {
NoOp(This fails);
};
};
//-example: Next line bracket
context failed7
{
1 => {
NoOp(This fails);
};
};
//-example: Switches and eswitches
context failed8 {
switches {
IAX2/abox;
};
/* Anything below fails */
1 => {
};
};
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Re: [asterisk-users] AEL2 Syntax Highlighting
On Fri, Oct 12, 2007 at 05:24:29PM -0500, Perssy Llamosas wrote: > Hi, > > I am looking for a syntax highlighter for AEL2. Google is not helping, > so I thought you guys could help me. > > I found this vim syntax highlighter for AEL but it doesn't help if you > want to code in AEL2: > http://vim.sourceforge.net/scripts/script.php?script_id=1900 How is AEL2 syntax different from AEL? Can you please give examples where the above fails for AEL2? (or for AEL, for that matter) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL2 Syntax Highlighting
Hi, I am looking for a syntax highlighter for AEL2. Google is not helping, so I thought you guys could help me. I found this vim syntax highlighter for AEL but it doesn't help if you want to code in AEL2: http://vim.sourceforge.net/scripts/script.php?script_id=1900 Cheers, PLL. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL2 Includes in Macro...
On Mon, 2007-06-04 at 09:19 -0700, Douglas Garstang wrote:
> > Where’s Steve Murphy when you need him? J
I'm right here! :)
> This doesn?t seem to work in AEL2?
> Macro foo(arg1) { ?..
> Includes {
>Hangup;
> }
> }
>
> The error is: File: /etc/asterisk/extensions.ael, Line 59, Cols: 5-12:
Error: > > syntax error, unexpected KW_INCLUDES, expecting '}'
>
> The same error does not occur when the includes is in a context.
> I need to have the ability to include my hangup routine in macros, as
> theoretically, a hang up could occur while asterisk is processing code
> from the macro. This is Asterisk 1.4.4
Doug--
This is a good point. So good, I created bug #9883 for this issue. I'll
look into it ASAP.
murf
--
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Software Developer
Digium
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[asterisk-users] AEL2 Includes in Macro...
Where's Steve Murphy when you need him? :-)
This doesn't seem to work in AEL2...
Macro foo(arg1) {
.
Includes {
Hangup;
}
}
The error is: File: /etc/asterisk/extensions.ael, Line 59, Cols: 5-12:
Error: syntax error, unexpected KW_INCLUDES, expecting '}'
The same error does not occur when the includes is in a context.
I need to have the ability to include my hangup routine in macros, as
theoretically, a hang up could occur while asterisk is processing code
from the macro.
This is Asterisk 1.4.4
Doug.
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[asterisk-users] AEL2 on Asterisk 1.2.4
Hey all, I am very interested in using AEL2 (don't want to upgrade to 1.4 to get it though), but am having some problems upgrading/patching my asterisk system. I am following the instructions on the wiki: http://www.voip-info.org/wiki/view/Asterisk+AEL2#AEL2AnnouncementsandNews But get the following error: "'http://svn.digium.com/svn/asterisk/team/murf/AEL2-1.2/diffs.AEL2.patch' refers to a file, not a directory" This refers to the process of including the patch as described in the first portion of the wiki page. Am am still new to linux so the problem could be just me, but I believe I followed the instructions. They are pretty simple after all. BTW, I tried both ways described and I could not get either to work. Thanks for any help, -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL2 Confusion
AEL2 is will be in 1.4 AEL2 is not in 1.2.x The Wiki was wrong. Bart Fisher wrote: I just downloaded and installed asterisk-1.2.13 Reading http://www.voip-info.org/wiki/view/Asterisk+AEL2 says I should be using AEL2, but what's downloaded is pbx_ael not pbx_ael2 as expected. Where and how do I get current release of AEL2 - Is there some 'How To' somewhere? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL2 Confusion
I just downloaded and installed asterisk-1.2.13 Reading http://www.voip-info.org/wiki/view/Asterisk+AEL2 says I should be using AEL2, but what's downloaded is pbx_ael not pbx_ael2 as expected. Where and how do I get current release of AEL2 - Is there some 'How To' somewhere? TIA Bart ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL2 in 1.2
Nevermind, seems scp'ing the source directory over didn't do me any good. On Fri, 2006-11-03 at 09:25 -0600, Aaron Daniel wrote: > I know I compiled AEL2 into 1.2 before, considering I just copied my > source from one server to another, yet I can't seem to figure out why > I'm getting this error. Anyone have any ideas? > > make[1]: Entering directory `/usr/local/src/asterisk-svn/asterisk/pbx' > flex argdesc.l > "argdesc.l", line 19: unrecognized %option: reentrant > "argdesc.l", line 20: unrecognized %option: bison-bridge > "argdesc.l", line 21: unrecognized %option: bison-locations > make[1]: *** [argdesc_lex.c] Error 1 > make[1]: Leaving directory `/usr/local/src/asterisk-svn/asterisk/pbx' > make: *** [subdirs] Error 1 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL2 in 1.2
I know I compiled AEL2 into 1.2 before, considering I just copied my source from one server to another, yet I can't seem to figure out why I'm getting this error. Anyone have any ideas? make[1]: Entering directory `/usr/local/src/asterisk-svn/asterisk/pbx' flex argdesc.l "argdesc.l", line 19: unrecognized %option: reentrant "argdesc.l", line 20: unrecognized %option: bison-bridge "argdesc.l", line 21: unrecognized %option: bison-locations make[1]: *** [argdesc_lex.c] Error 1 make[1]: Leaving directory `/usr/local/src/asterisk-svn/asterisk/pbx' make: *** [subdirs] Error 1 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL2 - CUT function usage
Hi,
In Asterisk 1.2.7, my AEL code looks like this:
macro callForwardHunt(numargs,numlist,typelist,ttr)
{
for(x=1;${x}<${numargs}+1;x=${x}+1)
{
CUT(number=numlist,-,${x});
CUT(type=typelist,-,${x});
NoOp(${number});
NoOp(${type});
Dial(${type}${number},${ttr});
};
};
In Asterisk 1.4.0beta3, the CUT function looks like this:
NoOp(${range});
Set(time_range=${CUT(range|/|1)});
NoOp(${time_range});
No I understand that the CUT application has been removed in 1.4, so now I am usung the CUT
function, but where is it explained that you have to have to use SET and the commas ',' has to be
replaced with '|'.
Or have I done something stupidly wrong :)
--
thanks,
yusuf
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[asterisk-users] AEL2 and the variables
Hi,
I am using Asterisk 1.2.12.1 + the AEL2 patch.
If I use a variable instead of the extension itself, an incoming call cannot
be connected.
${ID-FST1} => Dial(SIP/gs|15|r); <== NON ok
sip debug shows :
Looking for 6674262730 in interne (domain 192.168.1.14)
SIP/2.0 404 Not Found
Is it a bug or am I doing something wrong?
Thank you.
//===
// extensions.ael2
globals {
ID-FST1=6674262730;
GS=SIP/gs;
}
context entrant {
//6674262730 => Dial(SIP/gs,15,r); <== OK
${ID-FST1} => Dial(SIP/gs|15|r); <== NON ok
}
context interne {
includes {
entrant;
}
}
-
Dominique Dartois
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Re: [asterisk-users] AEL2 Catching on?
Rushowr wrote: Is it just me or am I seeing more AEL2 code in people's examples? Could If you're a C programmer, then yes. But, if you're like me, with very little programming skills; no. AEL to me, is too much like looking at C code. 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL2 Catching on?
Is it just me or am I seeing more AEL2 code in people's examples? Could it be that AEL2 is starting to finally catch on? SKM -AEL2 Fanatic, Potato Eater, and General Lurker ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL2 #include madness in Asterisk 1.4 - Murf?
Asterisk 1.4 beta2.
My top level /etc/asterisk/extensions.ael has the following two lines:
#include "include/syst/extensions.ael"
#include "include/btck/extensions.ael"
Here is the console output on Asterisk load.
app_system.so => (Generic System() application)
[Oct 4 15:48:15] NOTICE[1143]: pbx_ael.c:3798 pbx_load_module: Starting AEL
load process.
[Oct 4 15:48:15] NOTICE[1143]: pbx_ael.c:3805 pbx_load_module: AEL load
process: calculated config file name '/etc/asterisk/extensions.ael'.
[Oct 4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex: --Read in included
file /etc/asterisk/include/syst/extensions.ael, 4130 chars
[Oct 4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex: --Read in included
file /etc/asterisk/include/syst/macros.ael, 1463 chars
[Oct 4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex: --Read in included
file /etc/asterisk/include/syst/dundiapps.ael, 758 chars
[Oct 4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex: --Read in included
file /etc/asterisk/include/syst/rdapps.ael, 275 chars
[Oct 4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex: --Read in included
file /etc/asterisk/include/btck/extensions.ael, 1385 chars
[Oct 4 15:48:15] NOTICE[1143]: pbx_ael.c:3808 pbx_load_module: AEL load
process: parsed config file name '/etc/asterisk/extensions.ael'.
[Oct 4 15:48:15] ERROR[1143]: pbx_ael.c:1162 check_goto: Error: file
/etc/asterisk/include/syst/extensions.ael, line 157-157: goto: no label remote
exists in the current extension!
[Oct 4 15:48:15] ERROR[1143]: pbx_ael.c:1162 check_goto: Error: file
/etc/asterisk/include/syst/extensions.ael, line 159-159: goto: no label local
exists in the current extension!
[Oct 4 15:48:15] ERROR[1143]: pbx_ael.c:3821 pbx_load_module: Sorry, but 0
syntax errors and 2 semantic errors were detected. It doesn't make sense to
compile.
pbx_ael.so => (Asterisk Extension Language Compiler)
Here's the context from /etc/asterisk/include/syst/extensions.ael, that
contains lines 157 that the parser is complaining about:
148 context syst_Route {
149
150 _[*0123456789]. => {
151 NoOp(*** Originated call ${CALLERID} -> ${EXTEN});
152 Set(TMP=${CALLERID(number)});
153 &SysLogger(This is a test message);
154 &FastAGIConnectGet(CALLERID);
155 ChanIsAvail(SIP/${EXTEN});
156 if ("${AVAILCHAN}" = "") {
157 goto remote;
158 } else {
159 goto local;
160 }
161 remote:
162 NoOp(REMOTE);
163 Set(PATH=${DUNDILOOKUP(3254103,DUNDIRegistr)});
164 //Set(PATH=${DUNDILOOKUP(${EXTEN},DUNDIRegistr)});
165 Dial(${PATH});
166 Hangup();
167 local:
168 NoOp(LOCAL);
169 Dial(SIP/${EXTEN});
170 Hangup();
171
172 }
173 }
As you can quite clearly see, labels 'remote' and 'local' DO exist in the
syst_Route context.
Now, if I switcheroo the two includes around in the top level
/etc/asterisk/extensions.ael, to:
#include "include/btck/extensions.ael"
#include "include/syst/extensions.ael"
and reload Asterisk, I get:
[Oct 4 15:57:28] NOTICE[1202]: pbx_ael.c:3813 pbx_load_module: AEL load
process: compiled config file name '/etc/asterisk/extensions.ael'.
[Oct 4 15:57:28] NOTICE[1202]: pbx_ael.c:3816 pbx_load_module: AEL load
process: merged config file name '/etc/asterisk/extensions.ael'.
[Oct 4 15:57:28] WARNING[1202]: pbx.c:6194 ast_context_verify_includes:
Context 'syst_PSTNStart' tries includes nonexistent context 'syst_AppACDQueue'
[Oct 4 15:57:28] WARNING[1202]: pbx.c:6194 ast_context_verify_includes:
Context 'btck_CallStart' tries includes nonexistent context 'syst_ACD'
[Oct 4 15:57:28] NOTICE[1202]: pbx_ael.c:3819 pbx_load_module: AEL load
process: verified config file name '/etc/asterisk/extensions.ael'.
pbx_ael.so => (Asterisk Extension Language Compiler)
There are no errors about nonexistent labels in the syst_Route extension. I
would not have thought that #include order made any difference, since all we
are doing is pulling a bunch of contexts into a global context space.
Anyone? Mr Murpy, care to take a shot at it? :)
Doug.
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[asterisk-users] AEL2 patch for Asterisk 1.2.12.1
I want to try AEL2. This page: http://voip-info.org/wiki/view/Asterisk+AEL2 gives instructions to generate a patch from subversion: svn diff http://svn.digium.com/svn/asterisk/branches/1.2 http://svn.digium.com/svn/asterisk/team/murf/AEL2-1.2 > AEL.patch My doubt is: is "branches/1.2" the latest (i.e., exactly the same as "tags/1.2.12.1") or is it the original 1.2 version? Will the svn patch work if I do it under a 1.2.12.1 source tree using "branches/1.2" or should I use the "tags/1.2.12.1" URL? BarZ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL2 patch issues
I'm trying to patch the Asterisk 1.2 source to support AEL2 as follows: svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 cd 1.2 svn diff http://svn.digium.com/svn/asterisk/branches/1.2 http://svn.digium.com/svn/asterisk/team/murf/AEL2-1.2 > AEL.patch patch -p0 < AEL.patch make install The patch and build go fine, but I get the following errors when asterisk attempts to load the AEL2 module on startup: # [pbx_ael2.so]Aug 25 15:16:25 WARNING[28836]: loader.c:348 __load_resource: No usecount in module /usr/lib/asterisk/modules/pbx_ael2.so # Aug 25 15:16:25 WARNING[28836]: loader.c:380 __load_resource: 1 error loading module /usr/lib/asterisk/modules/pbx_ael2.so, aborted # Aug 25 15:16:25 WARNING[28836]: loader.c:554 load_modules: Loading module pbx_ael2.so failed! The box is running CentOS 4.3. Anybody else experiencing the same issues or am I doing something wrong? Thanks. Stephen Kratzer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] AEL2 Looping
Douglas,
Awesome! I don't know why I didn't get to the point of removing all the
spaces, probably got distracted by some shiny object ;-)
Anyway, thanks for the update!
Rushowr
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Saturday, July 29, 2006 6:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users
Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] AEL2 Looping
I actually did get it to work, by removing _all_ spaces from the for line...
for (x=0;${x}<3;x=${x}+1) {
This works for me. It's just a matter of finding WHICH space is breaking it.
-Original Message-
From: Rushowr [mailto:[EMAIL PROTECTED]
Sent: Sat 7/29/2006 12:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc:
Subject: RE: [asterisk-users] AEL2 Looping
>> context new_pbx_betty_start {
>>
>> _X. => {
>> for (x=0; ${x} < 3; x=${x} + 1) {
>> Verbose(x is ${x} !);
>> }
>> };
>>
>> }
>I would have to see the output of "show dialplan
new_pbx_betty_start" to
know exactly >what is going on. However, I'm guessing that if you
remove
the space between the
>semicolon and the "x=${x} + 1", it will work.
>On pretty much everything except expression evaluation (such as
${x} < 3),
Asterisk
>is sensitive to whitespace. " x=${x} + 1" was most likely
translated
directly into
>Set( x=$[${x} = 1]). That means you are setting the variable name,
" x",
including
>the leading space. That is not the same variable as ${x} which you
are
using
>everywhere else.
Russel,
Stupid question, but isn't the AEL2 parser supposed to handle the
above code
first? Hypothetically, if the parser DOES handle the code the
example given
by Murf on voip-info (which is the exact code Douglas posted, other
than the
name new_pbx_betty_start) should work properly. Also, to answer your
question, removing the space does not help. I'm actually getting a
bug
report together concerning this, and I tested it with and without
spaces in
multiple places in the for loop definition. I'd give examples but I
don't
have access right now.
Keep up the great work guys!
Rushowr
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RE: [asterisk-users] AEL2 Looping
I actually did get it to work, by removing _all_ spaces from the for line...
for (x=0;${x}<3;x=${x}+1) {
This works for me. It's just a matter of finding WHICH space is breaking it.
-Original Message-
From: Rushowr [mailto:[EMAIL PROTECTED]
Sent: Sat 7/29/2006 12:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc:
Subject: RE: [asterisk-users] AEL2 Looping
>> context new_pbx_betty_start {
>>
>> _X. => {
>> for (x=0; ${x} < 3; x=${x} + 1) {
>> Verbose(x is ${x} !);
>> }
>> };
>>
>> }
>I would have to see the output of "show dialplan new_pbx_betty_start"
to
know exactly >what is going on. However, I'm guessing that if you
remove
the space between the
>semicolon and the "x=${x} + 1", it will work.
>On pretty much everything except expression evaluation (such as ${x} <
3),
Asterisk
>is sensitive to whitespace. " x=${x} + 1" was most likely translated
directly into
>Set( x=$[${x} = 1]). That means you are setting the variable name, "
x",
including
>the leading space. That is not the same variable as ${x} which you are
using
>everywhere else.
Russel,
Stupid question, but isn't the AEL2 parser supposed to handle the above
code
first? Hypothetically, if the parser DOES handle the code the example
given
by Murf on voip-info (which is the exact code Douglas posted, other
than the
name new_pbx_betty_start) should work properly. Also, to answer your
question, removing the space does not help. I'm actually getting a bug
report together concerning this, and I tested it with and without
spaces in
multiple places in the for loop definition. I'd give examples but I
don't
have access right now.
Keep up the great work guys!
Rushowr
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RE: [asterisk-users] AEL2 Looping
>> context new_pbx_betty_start {
>>
>> _X. => {
>> for (x=0; ${x} < 3; x=${x} + 1) {
>> Verbose(x is ${x} !);
>> }
>> };
>>
>> }
>I would have to see the output of "show dialplan new_pbx_betty_start" to
know exactly >what is going on. However, I'm guessing that if you remove
the space between the
>semicolon and the "x=${x} + 1", it will work.
>On pretty much everything except expression evaluation (such as ${x} < 3),
Asterisk
>is sensitive to whitespace. " x=${x} + 1" was most likely translated
directly into
>Set( x=$[${x} = 1]). That means you are setting the variable name, " x",
including
>the leading space. That is not the same variable as ${x} which you are
using
>everywhere else.
Russel,
Stupid question, but isn't the AEL2 parser supposed to handle the above code
first? Hypothetically, if the parser DOES handle the code the example given
by Murf on voip-info (which is the exact code Douglas posted, other than the
name new_pbx_betty_start) should work properly. Also, to answer your
question, removing the space does not help. I'm actually getting a bug
report together concerning this, and I tested it with and without spaces in
multiple places in the for loop definition. I'd give examples but I don't
have access right now.
Keep up the great work guys!
Rushowr
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Re: [asterisk-users] AEL2 Looping
- Douglas Garstang <[EMAIL PROTECTED]> wrote:
> context new_pbx_betty_start {
>
> _X. => {
> for (x=0; ${x} < 3; x=${x} + 1) {
> Verbose(x is ${x} !);
> }
> };
>
> }
>
> Here's the output.
>
> The var x never gets incremented! Is this a bug?
> The while loops seem to work ok.
I would have to see the output of "show dialplan new_pbx_betty_start" to know
exactly what is going on. However, I'm guessing that if you remove the space
between the semicolon and the "x=${x} + 1", it will work.
On pretty much everything except expression evaluation (such as ${x} < 3),
Asterisk is sensitive to whitespace. " x=${x} + 1" was most likely translated
directly into Set( x=$[${x} = 1]). That means you are setting the variable
name, " x", including the leading space. That is not the same variable as ${x}
which you are using everywhere else.
--
Russell Bryant
Software Developer
Digium, Inc.
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[asterisk-users] AEL2 Looping
I was just experimenting with AEL2, and tried to use a for loop as per the
example.
Here's what I have.
context new_pbx_betty_start {
_X. => {
for (x=0; ${x} < 3; x=${x} + 1) {
Verbose(x is ${x} !);
}
};
}
Here's the output.
The var x never gets incremented! Is this a bug?
The while loops seem to work ok.
x is 0 !
-- Executing Set("SIP/3254101-db4b", " x=1") in new stack
-- Executing Goto("SIP/3254101-db4b", "2") in new stack
-- Goto (new_pbx_betty_start,36521478,2)
-- Executing GotoIf("SIP/3254101-db4b", "1?3:6") in new stack
-- Goto (new_pbx_betty_start,36521478,3)
-- Executing Verbose("SIP/3254101-db4b", "x is 0 !") in new stack
x is 0 !
-- Executing Set("SIP/3254101-db4b", " x=1") in new stack
-- Executing Goto("SIP/3254101-db4b", "2") in new stack
-- Goto (new_pbx_betty_start,36521478,2)
-- Executing GotoIf("SIP/3254101-db4b", "1?3:6") in new stack
-- Goto (new_pbx_betty_start,36521478,3)
-- Executing Verbose("SIP/3254101-db4b", "x is 0 !") in new stack
x is 0 !
-- Executing Set("SIP/3254101-db4b", " x=1") in new stack
-- Executing Goto("SIP/3254101-db4b", "2") in new stack
-- Goto (new_pbx_betty_start,36521478,2)
-- Executing GotoIf("SIP/3254101-db4b", "1?3:6") in new stack
-- Goto (new_pbx_betty_start,36521478,3)
-- Executing Verbose("SIP/3254101-db4b", "x is 0 !") in new stack
x is 0 !
-- Executing Set("SIP/3254101-db4b", " x=1") in new stack
-- Executing Goto("SIP/3254101-db4b", "2") in new stack
-- Goto (new_pbx_betty_start,36521478,2)
-- Executing GotoIf("SIP/3254101-db4b", "1?3:6") in new stack
-- Goto (new_pbx_betty_start,36521478,3)
-- Executing Verbose("SIP/3254101-db4b", "x is 0 !") in new stack
x is 0 !
-- Executing Set("SIP/3254101-db4b", " x=1") in new stack
-- Executing Goto("SIP/3254101-db4b", "2") in new stack
-- Goto (new_pbx_betty_start,36521478,2)
-- Executing GotoIf("SIP/3254101-db4b", "1?3:6") in new stack
-- Goto (new_pbx_betty_start,36521478,3)
-- Executing Verbose("SIP/3254101-db4b", "x is 0 !") in new stack
x is 0 !
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Re: [Asterisk-Users] AEL2
Hello All , On Fri, 9 Jun 2006, Gonzalo Servat wrote: On 6/9/06, Joshua Colp <[EMAIL PROTECTED]> wrote: [..snip..] I'd just like to note that AEL2 was brought over into Asterisk trunk (what will become 1.4) and the old AEL removed. That's where most development is taking place on AEL2, and why you don't see patches on the bug tracker. Hi Joshua, I was just reading the bug report and noticed it has been merged. Awesome news! I'm still using 1.2.x so sticking to AEL for now, but I'm going to quickly move to AEL2 as soon as I upgrade to 1.4! (whenever it comes out) Regards, Gonzalo. Something along the same lines . Does anyone have link to complete documentation of AEL2 ? Did a small bit of checking & nothing that seemed complete . Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3600 14th Ave SE #20-103 | Give me Linux | | [EMAIL PROTECTED] | Olympia , WA. 98501 | only on AXP | +--+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AEL2
On 6/9/06, Joshua Colp <[EMAIL PROTECTED]> wrote: [..snip..] I'd just like to note that AEL2 was brought over into Asterisk trunk (what will become 1.4) and the old AEL removed. That's where most development is taking place on AEL2, and why you don't see patches on the bug tracker. Hi Joshua, I was just reading the bug report and noticed it has been merged. Awesome news! I'm still using 1.2.x so sticking to AEL for now, but I'm going to quickly move to AEL2 as soon as I upgrade to 1.4! (whenever it comes out) Regards, Gonzalo. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AEL2
Gonzalo Servat wrote: Hi Doug, On 6/9/06, Doug Crompton <[EMAIL PROTECTED]> wrote: Replying to myself here... I got the latest 1.2 head via svn. Did a patch diff'ed it to latest AEL2 (as described at: http://voip-info.org/wiki/view/Asterisk+AEL2 Patched it. All went fine. On compile I get the following error [..snip..] This probably won't help you much, but you'll find one big bug report in the Digium Bug Database used for AEL2 where you'll see lots of input from the AEL2 guy. He's a good guy and real helpful. Infact, here's the URL: http://bugs.digium.com/view.php?id=6021 Hope this helps! Regards, Gonzalo. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I'd just like to note that AEL2 was brought over into Asterisk trunk (what will become 1.4) and the old AEL removed. That's where most development is taking place on AEL2, and why you don't see patches on the bug tracker. -- Joshua Colp Software Developer Digium P - 256-428-6066 C - 506-878-0147 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AEL2
Hi Doug, On 6/9/06, Doug Crompton <[EMAIL PROTECTED]> wrote: Replying to myself here... I got the latest 1.2 head via svn. Did a patch diff'ed it to latest AEL2 (as described at: http://voip-info.org/wiki/view/Asterisk+AEL2 Patched it. All went fine. On compile I get the following error [..snip..] This probably won't help you much, but you'll find one big bug report in the Digium Bug Database used for AEL2 where you'll see lots of input from the AEL2 guy. He's a good guy and real helpful. Infact, here's the URL: http://bugs.digium.com/view.php?id=6021 Hope this helps! Regards, Gonzalo. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AEL2
Replying to myself here... I got the latest 1.2 head via svn. Did a patch diff'ed it to latest AEL2 (as described at: http://voip-info.org/wiki/view/Asterisk+AEL2 Patched it. All went fine. On compile I get the following error pbx_ael2.c: In function `check_goto': pbx_ael2.c:1063: parse error before `struct' pbx_ael2.c:1065: `x3' undeclared (first use in this function) pbx_ael2.c:1065: (Each undeclared identifier is reported only once pbx_ael2.c:1065: for each function it appears in.) pbx_ael2.c: In function `find_label_in_current_context': pbx_ael2.c:1532: parse error before `struct' pbx_ael2.c:1534: `x3' undeclared (first use in this function) make[1]: *** [pbx_ael2.o] Error 1 Which boils down to a definition of x3 in pbx_ael2.c Anyone played with this? Doug On Thu, 8 Jun 2006, Doug Crompton wrote: > Being rather new to Asterisk I was wondering what the current status of > AEL2 is? I see reference to it back in January but that was many versions > ago. Is it in the current code? > > Doug > > > * Doug Crompton * > * Richboro, PA 18954* > * 215-431-6307 * > ** > * [EMAIL PROTECTED]* > * http://www.crompton.com * > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > "Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety." -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AEL2
Being rather new to Asterisk I was wondering what the current status of AEL2 is? I see reference to it back in January but that was many versions ago. Is it in the current code? Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AEL2 and CID
Correct me if I'm wrong, but doing this CID stuff in AEL may not make as
much sense in terms of converting dialplans over as it seems. I say this,
because with the original usage of the CID checking in the old extension
language, you could base PRIORITIES on the CID, therefore changing only
part of the actual extension logic. With how you're looking at it, that
would effectively render an extension into two separate logical forms,
which I know would definitely confuse people when converting the
languages. Also, doing the CID checks inside the extension gives a larger
degree of control, and makes the dialplan a bit more eligible. From an
administration standpoint, you could have multiple EXTEN/CID's strewn
about, but if you strictly use the in-extension checking, you know that
*THIS* is the extension you're looking for, and *THOSE* CID's are the ones
that are going to do something different.
Just my thoughts on the matter :) Hope they help a little.
On Thu, 1 Jun 2006, Julian Lyndon-Smith wrote:
Yes! That's the answer I was hoping ! I'm not stupid - it's a *feature* :)
Anything you need testing, let me know !
Julian
Steve Murphy wrote:
From: Douglas Garstang <[EMAIL PROTECTED]>
Yikes! I'm glad I didn't take the plunge into AEL2. Get #include
functionality, but lose cid in the dialplan. Hmmm.
-Original Message-
From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 31, 2006 1:21 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AEL2 and CID
Does anyone know how to get CID working in AEL2 ?
In extensions.conf you can do:
exten => 111/666,1,PlayBack(demo-congrats)
exten => 111/666,2,Hangup()
exten => 111,1,PlayBack(demo-moreinfo)
exten => 111,2,Hangup()
and if callerid 666 dialed 111, they would get demo-congrats, everyone
else gets demo-moreinfo.
In AEL:
111 => {
Playback(demo-moreinfo);
Hangup();
};
111/666 => {
Playback(demo-congrats);
Hangup();
};
does not work. It always plays demo-moreinfo.
I cannot find and docs on how to do this.
Anyone got any idea ?
Many thanks.
Julian
Douglas!! Now, now!! That is not the proper attitude!! ;^) AEL is still
in the formation, we'll look to see if this little problem can be
remedied.
murf
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--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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Re: [Asterisk-Users] AEL2 and CID
Yes! That's the answer I was hoping ! I'm not stupid - it's a *feature* :)
Anything you need testing, let me know !
Julian
Steve Murphy wrote:
From: Douglas Garstang <[EMAIL PROTECTED]>
Yikes! I'm glad I didn't take the plunge into AEL2. Get #include
functionality, but lose cid in the dialplan. Hmmm.
-Original Message-
From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 31, 2006 1:21 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AEL2 and CID
Does anyone know how to get CID working in AEL2 ?
In extensions.conf you can do:
exten => 111/666,1,PlayBack(demo-congrats)
exten => 111/666,2,Hangup()
exten => 111,1,PlayBack(demo-moreinfo)
exten => 111,2,Hangup()
and if callerid 666 dialed 111, they would get demo-congrats,
everyone
else gets demo-moreinfo.
In AEL:
111 => {
Playback(demo-moreinfo);
Hangup();
};
111/666 => {
Playback(demo-congrats);
Hangup();
};
does not work. It always plays demo-moreinfo.
I cannot find and docs on how to do this.
Anyone got any idea ?
Many thanks.
Julian
Douglas!! Now, now!! That is not the proper attitude!! ;^) AEL is still
in the formation, we'll look to see if this little problem can be
remedied.
murf
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RE: [Asterisk-Users] AEL2 and CID
> From: Douglas Garstang <[EMAIL PROTECTED]>
> Yikes! I'm glad I didn't take the plunge into AEL2. Get #include
> functionality, but lose cid in the dialplan. Hmmm.
>
> > -Original Message-
> > From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
> > Sent: Wednesday, May 31, 2006 1:21 PM
> > To: asterisk-users@lists.digium.com
> > Subject: [Asterisk-Users] AEL2 and CID
> >
> >
> > Does anyone know how to get CID working in AEL2 ?
> >
> > In extensions.conf you can do:
> >
> > exten => 111/666,1,PlayBack(demo-congrats)
> > exten => 111/666,2,Hangup()
> >
> > exten => 111,1,PlayBack(demo-moreinfo)
> > exten => 111,2,Hangup()
> >
> > and if callerid 666 dialed 111, they would get demo-congrats,
> > everyone
> > else gets demo-moreinfo.
> >
> > In AEL:
> >
> > 111 => {
> > Playback(demo-moreinfo);
> > Hangup();
> > };
> >
> > 111/666 => {
> > Playback(demo-congrats);
> > Hangup();
> > };
> >
> > does not work. It always plays demo-moreinfo.
> >
> > I cannot find and docs on how to do this.
> >
> > Anyone got any idea ?
> >
> > Many thanks.
> >
> > Julian
> >
Douglas!! Now, now!! That is not the proper attitude!! ;^) AEL is still
in the formation, we'll look to see if this little problem can be
remedied.
murf
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Re: [Asterisk-Users] AEL2 and CID
Yeah, thanks, that was the way I was leaning to. Just was wanting to
know if it was a syntax I was getting wrong, or if there is no other way
of doing this.
Julian.
Mojo with Horan & Company, LLC wrote:
I guess you could do this, but it would be a little cumbersome:
context incoming {
s => {
if ("${CALLERID(num)}" = "8005551212")
{
NoOp("Dir. Asst. calling");
}
else if ("${CALLERID(num)}" = "800444")
{
NoOp("ANI calling");
}
}
}
Douglas Garstang wrote:
Yikes! I'm glad I didn't take the plunge into AEL2. Get #include
functionality, but lose cid in the dialplan. Hmmm.
-Original Message-
From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 31, 2006 1:21 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AEL2 and CID
Does anyone know how to get CID working in AEL2 ?
In extensions.conf you can do:
exten => 111/666,1,PlayBack(demo-congrats)
exten => 111/666,2,Hangup()
exten => 111,1,PlayBack(demo-moreinfo)
exten => 111,2,Hangup()
and if callerid 666 dialed 111, they would get demo-congrats,
everyone else gets demo-moreinfo.
In AEL:
111 => {
Playback(demo-moreinfo);
Hangup();
};
111/666 => {
Playback(demo-congrats);
Hangup();
};
does not work. It always plays demo-moreinfo.
I cannot find and docs on how to do this.
Anyone got any idea ?
Many thanks.
Julian
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Re: [Asterisk-Users] AEL2 and CID
Of course, without the quotes in the NoOp parameters, if it matters:
---
...
NoOp(Dir. Asst. calling);
...
Douglas Garstang wrote:
Yikes! I'm glad I didn't take the plunge into AEL2. Get #include functionality,
but lose cid in the dialplan. Hmmm.
-Original Message-
From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 31, 2006 1:21 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AEL2 and CID
Does anyone know how to get CID working in AEL2 ?
In extensions.conf you can do:
exten => 111/666,1,PlayBack(demo-congrats)
exten => 111/666,2,Hangup()
exten => 111,1,PlayBack(demo-moreinfo)
exten => 111,2,Hangup()
and if callerid 666 dialed 111, they would get demo-congrats,
everyone
else gets demo-moreinfo.
In AEL:
111 => {
Playback(demo-moreinfo);
Hangup();
};
111/666 => {
Playback(demo-congrats);
Hangup();
};
does not work. It always plays demo-moreinfo.
I cannot find and docs on how to do this.
Anyone got any idea ?
Many thanks.
Julian
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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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Re: [Asterisk-Users] AEL2 and CID
I guess you could do this, but it would be a little cumbersome:
context incoming {
s => {
if ("${CALLERID(num)}" = "8005551212")
{
NoOp("Dir. Asst. calling");
}
else if ("${CALLERID(num)}" = "800444")
{
NoOp("ANI calling");
}
}
}
Douglas Garstang wrote:
Yikes! I'm glad I didn't take the plunge into AEL2. Get #include functionality,
but lose cid in the dialplan. Hmmm.
-Original Message-
From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 31, 2006 1:21 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AEL2 and CID
Does anyone know how to get CID working in AEL2 ?
In extensions.conf you can do:
exten => 111/666,1,PlayBack(demo-congrats)
exten => 111/666,2,Hangup()
exten => 111,1,PlayBack(demo-moreinfo)
exten => 111,2,Hangup()
and if callerid 666 dialed 111, they would get demo-congrats,
everyone
else gets demo-moreinfo.
In AEL:
111 => {
Playback(demo-moreinfo);
Hangup();
};
111/666 => {
Playback(demo-congrats);
Hangup();
};
does not work. It always plays demo-moreinfo.
I cannot find and docs on how to do this.
Anyone got any idea ?
Many thanks.
Julian
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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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RE: [Asterisk-Users] AEL2 and CID
Yikes! I'm glad I didn't take the plunge into AEL2. Get #include functionality,
but lose cid in the dialplan. Hmmm.
> -Original Message-
> From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, May 31, 2006 1:21 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] AEL2 and CID
>
>
> Does anyone know how to get CID working in AEL2 ?
>
> In extensions.conf you can do:
>
> exten => 111/666,1,PlayBack(demo-congrats)
> exten => 111/666,2,Hangup()
>
> exten => 111,1,PlayBack(demo-moreinfo)
> exten => 111,2,Hangup()
>
> and if callerid 666 dialed 111, they would get demo-congrats,
> everyone
> else gets demo-moreinfo.
>
> In AEL:
>
> 111 => {
> Playback(demo-moreinfo);
> Hangup();
> };
>
> 111/666 => {
> Playback(demo-congrats);
> Hangup();
> };
>
> does not work. It always plays demo-moreinfo.
>
> I cannot find and docs on how to do this.
>
> Anyone got any idea ?
>
> Many thanks.
>
> Julian
>
>
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[Asterisk-Users] AEL2 and CID
Does anyone know how to get CID working in AEL2 ?
In extensions.conf you can do:
exten => 111/666,1,PlayBack(demo-congrats)
exten => 111/666,2,Hangup()
exten => 111,1,PlayBack(demo-moreinfo)
exten => 111,2,Hangup()
and if callerid 666 dialed 111, they would get demo-congrats, everyone
else gets demo-moreinfo.
In AEL:
111 => {
Playback(demo-moreinfo);
Hangup();
};
111/666 => {
Playback(demo-congrats);
Hangup();
};
does not work. It always plays demo-moreinfo.
I cannot find and docs on how to do this.
Anyone got any idea ?
Many thanks.
Julian
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RE: [Asterisk-Users] AEL2 -- The Future --
On Fri, 2006-01-13 at 10:24 -0700, Douglas Garstang wrote: > Does ael support #include statements yet? > Is there any way to perform a reload in asterisk and reload extensions.ael? > > If both of these aren't available yet, then AEL isn't ready for real-world > use. > > Doug. > Oh, and one more thing about the reload of AEL2--- I provide in the patch, in the utils/ subdir, the standalone exec, aelparse, which will parse the file and generate any errors it might have during the load. It is far better to lint out the problems before feeding it to asterisk, and taking the whole switch out of commission whilst you muck with the file. There are a bunch of errors I just plain will not be able to detect before run-time. Hopefully you don't have to use reload until all the syntax and semantic errors are cleaned up. murf -- Steve Murphy <[EMAIL PROTECTED]> Electronic Tools Company ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AEL2 -- The Future --
On Fri, 2006-01-13 at 11:42 -0700, Steve Murphy wrote: > to reload, > > > ael2 reload > ael1 supported that too, the file inclusion was a really nice feature to add, something I felt was fundamentally missing from ael1. I like the syntax checker too, so you dont check by potentially breaking a live install with typos :/ Good work so far :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AEL2 -- The Future --
On Fri, 2006-01-13 at 10:24 -0700, Douglas Garstang wrote: > Does ael support #include statements yet? > Is there any way to perform a reload in asterisk and reload extensions.ael? > > If both of these aren't available yet, then AEL isn't ready for real-world > use. > > Doug. > Doug-- Good news, then! #include "" where can be either absolute, or relative to /etc/asterisk and, on top of that, the #include "" stuff can be used to include contexts, extensions, or even just stuff inside an extension. You can use it ALMOST anywhere in the file. The scanner replaces it with the contents of the file. Nesting is OK. 50 levels deep max. to reload, > ael2 reload in the asterisk command shell. This is all there in the WIKI. Do not confuse AEL with AEL2. See http://www.voip-info.org/wiki/view/Asterisk+AEL2 AEL2, I've done in the last month. Check it out, work it over. Report your experience. murf -- Steve Murphy <[EMAIL PROTECTED]> Electronic Tools Company smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AEL2 -- The Future --
Douglas Garstang wrote: Does ael support #include statements yet? Based on http://www.voip-info.org/wiki/view/Asterisk+AEL2: To reload extensions.ael2, the following command can be issued at the CLI. *CLI> ael2 reload Is there any way to perform a reload in asterisk and reload extensions.ael? From the same site: You can include other files with the #include "filepath" construct. 1. include "/etc/asterisk/testfor.ael2" Read the site for more details. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AEL2 -- The Future --
Does ael support #include statements yet? Is there any way to perform a reload in asterisk and reload extensions.ael? If both of these aren't available yet, then AEL isn't ready for real-world use. Doug. -Original Message- From: Steve Murphy [mailto:[EMAIL PROTECTED] Sent: Friday, January 13, 2006 9:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] AEL2 -- The Future -- Call to Action! For those who have the courage/ability, go grab an SVN copy of the asterisk release, the HEAD version, and my latest patch, from: http://bugs.digium.com/view.php?id=6021 Right now, the latest version of the patch is 0.10. apply it to the SVN head version, and do a "make". Read the Wiki on AEL2: http://www.voip-info.org/wiki/view/Asterisk+AEL2 Look at the examples at: http://www.voip-info.org/wiki/view/AEL+Example+Snippets Then, sit down and rewrite your extension.conf to /etc/asterisk/extensions.ael2 Use "utils/aelparse -n" to check your file. Get rid of all the syntax errors. Repeat until clean. Then, see if your extensions.ael2 loads. Remove all the contexts except [general] from your extensions.conf, and restart asterisk. Test the new dialplan. Now, at this point, you have some information that would be useful to me! I need to know your trials, troubles, confusions, and solutions. Perhaps there's some added check that AEL2 could make that might have warned or helped you. Perhaps you'll help find some bug and solify AEL2. WHY ON EARTH WOULD I WANT TO DO ALL THAT WORK?, you might ask! In answer to that, my reply is the Parable of the Programmer: ** In the beginning, there was machine code, and the Programmer thought it was great. Then, along came the assembler, and the programmer found it very useful. Then, along came the macro assembler, and the programmer was excited indeed. But then came the "programming language", and the programmer left behind the "macro assembler", and never went back to it. ** Now, the goal is for the same to happen to you, in moving from the extensions.conf method of programming dialplans to extensions.ael2. If you don't think it will improve the quality of the code you write for dialplans, or reduce your costs of dialplan development, then we'll scrap the project and hand ourselves over to public humiliation. ;^) murf -- Steve Murphy <[EMAIL PROTECTED]> Electronic Tools Company ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AEL2 -- The Future --
Call to Action! For those who have the courage/ability, go grab an SVN copy of the asterisk release, the HEAD version, and my latest patch, from: http://bugs.digium.com/view.php?id=6021 Right now, the latest version of the patch is 0.10. apply it to the SVN head version, and do a "make". Read the Wiki on AEL2: http://www.voip-info.org/wiki/view/Asterisk+AEL2 Look at the examples at: http://www.voip-info.org/wiki/view/AEL+Example+Snippets Then, sit down and rewrite your extension.conf to /etc/asterisk/extensions.ael2 Use "utils/aelparse -n" to check your file. Get rid of all the syntax errors. Repeat until clean. Then, see if your extensions.ael2 loads. Remove all the contexts except [general] from your extensions.conf, and restart asterisk. Test the new dialplan. Now, at this point, you have some information that would be useful to me! I need to know your trials, troubles, confusions, and solutions. Perhaps there's some added check that AEL2 could make that might have warned or helped you. Perhaps you'll help find some bug and solify AEL2. WHY ON EARTH WOULD I WANT TO DO ALL THAT WORK?, you might ask! In answer to that, my reply is the Parable of the Programmer: ** In the beginning, there was machine code, and the Programmer thought it was great. Then, along came the assembler, and the programmer found it very useful. Then, along came the macro assembler, and the programmer was excited indeed. But then came the "programming language", and the programmer left behind the "macro assembler", and never went back to it. ** Now, the goal is for the same to happen to you, in moving from the extensions.conf method of programming dialplans to extensions.ael2. If you don't think it will improve the quality of the code you write for dialplans, or reduce your costs of dialplan development, then we'll scrap the project and hand ourselves over to public humiliation. ;^) murf -- Steve Murphy <[EMAIL PROTECTED]> Electronic Tools Company ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
