Re: [asterisk-users] ** in extensions.conf

[Tech Support]

Is ** also defined in features.conf?
Thanks;
John

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Wednesday, April 26, 2017 05:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ** in extensions.conf

On Wed, 26 Apr 2017, Jerry Geis wrote:

> dialplan show testing-sip
>   '**' =>   1. Noop(Testing)  
> [pbx_config]
> 2. Playback(demo-congrats)
> [pbx_config]
> 
> Looks like its there.
> 
> if I do ** "Dial" it works, but if I do "New Call" ** then "Dial" it 
> does not work. Weird. How do I get it to work for both cases. (glad I 
> tried the other)

I never use 'New Call' -- just 'Dial' and 'Redial,'

I suspect you'll need to fiddle with the Polycom dialplan. As soon as I press 
the first '*' my Poly sends the INVITE.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 https://www.linkedin.com/in/steve-edwards-4244281


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Re: [asterisk-users] ** in extensions.conf

[Steve Davies]

On Wed, 26 Apr 2017 at 20:29 Jerry Geis  wrote:

> I just tried this in my extensions.conf
>
> exten => **,1,Noop(Testing)
> exten => **,n,Playback(demo-congrats)
>
> Did a reload... and the above does not happen.
> I created as 12 instead of the ** and that works fine.
>
> Is there anyway to get the ** to work?  I also am using a polycom phone if
> that affects things. I'm using asterisk 13.15.0
>
> Thanks
>
> Jerry
>
>
On a Polycom handset, dialling '**' will by default be translated into '+'
before it is dialled. You could:

1) dial *..pause..* which will overcome that AFAIK.
2) Configure call.InternationalDialing.enabled="0" on the handset to stop
it.
3) Put a pattern of _[+],1,... into your dialplan.

That would be by guess anyway :)
Steve
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Re: [asterisk-users] ** in extensions.conf

[Steve Edwards]

On Wed, 26 Apr 2017, Jerry Geis wrote:


dialplan show testing-sip
  '**' =>           1. Noop(Testing)                              [pbx_config]
                    2. Playback(demo-congrats)                    [pbx_config]

Looks like its there.

if I do ** "Dial" it works, but if I do "New Call" ** then "Dial" it 
does not work. Weird. How do I get it to work for both cases. (glad I 
tried the other)


I never use 'New Call' -- just 'Dial' and 'Redial,'

I suspect you'll need to fiddle with the Polycom dialplan. As soon as I 
press the first '*' my Poly sends the INVITE.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281-- 
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Re: [asterisk-users] ** in extensions.conf

[Jerry Geis]

dialplan show testing-sip
  '**' =>   1. Noop(Testing)
 [pbx_config]
2. Playback(demo-congrats)
 [pbx_config]

Looks like its there.

if I do ** "Dial" it works, but if I do "New Call" ** then "Dial" it does
not work. Weird.
How do I get it to work for both cases. (glad I tried the other)

Jerry
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Re: [asterisk-users] ** in extensions.conf

[Steve Edwards]

On Wed, 26 Apr 2017, Jerry Geis wrote:


I just tried this in my extensions.conf

exten => **,1,Noop(Testing)
exten => **,n,Playback(demo-congrats)

Did a reload... and the above does not happen.
I created as 12 instead of the ** and that works fine.

Is there anyway to get the ** to work?  I also am using a polycom phone 
if that affects things. I'm using asterisk 13.15.0


Coincidentally, this is exactly how I exercise test code:

; test something
; (changes frequently)
exten = **,1,   verbose(1,[${EXTEN}@${CONTEXT}!${ANI}])
same = n,   answer()

I use an ancient Polycom IP 501 just fine.

Does 'dialplan show **@' yield any clues?

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281-- 
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Re: [asterisk-users] ** in extensions.conf

[Richard Mudgett]

On Wed, Apr 26, 2017 at 2:28 PM, Jerry Geis  wrote:

> I just tried this in my extensions.conf
>
> exten => **,1,Noop(Testing)
> exten => **,n,Playback(demo-congrats)
>
> Did a reload... and the above does not happen.
> I created as 12 instead of the ** and that works fine.
>
> Is there anyway to get the ** to work?  I also am using a polycom phone if
> that affects things. I'm using asterisk 13.15.0
>

A ** extension should work just fine.  I expect it is the dialplan in the
polycom phone
that doesn't allow it.

Richard
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[asterisk-users] ** in extensions.conf

[Jerry Geis]

I just tried this in my extensions.conf

exten => **,1,Noop(Testing)
exten => **,n,Playback(demo-congrats)

Did a reload... and the above does not happen.
I created as 12 instead of the ** and that works fine.

Is there anyway to get the ** to work?  I also am using a polycom phone if
that affects things. I'm using asterisk 13.15.0

Thanks

Jerry
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Re: [asterisk-users] recreating extensions.conf from live dialplan ?

[A J Stiles]

On Wednesday 13 Apr 2016, Jeremy Kister wrote:
> On 4/13/16 11:57 AM, A J Stiles wrote:
> > You could try
> > *CLI> dialplan show
> 
> Between my older backup and dialplan show, I guess that's my best shot.
> 
> Thanks :D

I'll have a go this lunchtime at knocking up a Perl script  {for that is my 
language of choice}  to try to recreate an extensions.conf file from the 
`dialplan show` CLI output.  All the necessary stuff seems to be there, even 
labels for GoTo statements .

-- 
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] recreating extensions.conf from live dialplan ?

[Jeremy Kister]

On 4/13/2016 1:26 PM, Steve Edwards wrote:

This should get you close:

  sudo asterisk -r -x 'dialplan show' >extensions.wip

and then feed extensions.wip through:


Ya, that's pretty good!  besides the fact that I've never used "same" (i 
understand where it's coming from) and a few contexts confuzzled 
(missing general/globals and extra parkedcalls - but again I get it) - 
it seems to be perfect.


One for a wiki, somewhere.


thanks,

--

Jeremy Kister
http://jeremy.kister.net/

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Re: [asterisk-users] recreating extensions.conf from live dialplan ?

[Steve Edwards]

On Wed, 13 Apr 2016, Steve Edwards wrote:


// label
if  ('[' == substr($line, 5, 1))
{
$line = str_replace(']', '[', $line);
$tokens = explode('[', $line);
printf("\tsame = n(%s),\t\t\t%s", $tokens[1], $line);
continue;
}


Damn. Missed a line of code.

Should be:

// label
if  ('[' == substr($line, 5, 1))
{
$line = str_replace(']', '[', $line);
$tokens = explode('[', $line);
$line = substr($line, strpos($line, ' ', 20) + 1, 255);
printf("\tsame = n(%s),\t\t\t%s", $tokens[1], $line);
continue;
}

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] recreating extensions.conf from live dialplan ?

[Steve Edwards]

On Wed, 13 Apr 2016, Jeremy Kister wrote:


Between my older backup and dialplan show, I guess that's my best shot.


This should get you close:

sudo asterisk -r -x 'dialplan show' >extensions.wip

and then feed extensions.wip through:

#!/usr/bin/env php


(I'm not all that hot of a PHP programmer, so if anybody has some 
constructive criticism, please jump it.)


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] recreating extensions.conf from live dialplan ?

[Steve Edwards]

On Wed, 13 Apr 2016, Steve Edwards wrote:


Will 'dialplan save' help?


I just tried this one. It writes the dialplan, but without the application 
arguements. Worthless.


Aside from just a great way to eff up your day, does 'dialplan save' have 
any value?


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] recreating extensions.conf from live dialplan ?

[Jeremy Kister]

On 4/13/16 11:57 AM, A J Stiles wrote:

You could try
*CLI> dialplan show


Between my older backup and dialplan show, I guess that's my best shot.

Thanks :D




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Re: [asterisk-users] recreating extensions.conf from live dialplan ?

[Jeremy Kister]

On 4/13/16 11:37 AM, Steve Edwards wrote:

Will 'dialplan save' help?


I just tried this one. It writes the dialplan, but without the
application arguements. Worthless.


right, was a good shot.  in my case I have writeprotect=yes in general, 
so that would have been the first hurdle.  but asterisk does have my 
latest-and-greatest code in memory and active in it's dialplan.  hoping 
for something similar...




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Re: [asterisk-users] recreating extensions.conf from live dialplan ?

[A J Stiles]

On Wednesday 13 Apr 2016, Jeremy Kister wrote:
> with the slip of a finger, i destroyed by extensions.conf (grep -i >
> extensions.conf)
> 
> I have a backup that is dozens of hours of code old.
> 
> is there a way i can use the asterisk cli (or some other asterisky
> method) to recreate that extensions.conf ?

You could try
*CLI> dialplan show
The output from this is not the same format as an extensions.conf file, but it 
will have all the relevant information in it; and it seems to include all the 
relevant infiormation.  So it could be made to look like an extensions.conf 
file, if you really have nothing else -- desperate situations call for 
desperate remedies, and all that .

-- 
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] recreating extensions.conf from live dialplan ?

[Steve Edwards]

On Wed, 13 Apr 2016, Jeremy Kister wrote:

is there a way i can use the asterisk cli (or some other asterisky method) to 
recreate that extensions.conf ?


sudo asterisk -r -x 'dialplan show' >extensions.wip

And then start cobbling up a script to parse and re-write into a usable 
format.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] recreating extensions.conf from live dialplan ?

[Steve Edwards]

On Wed, 13 Apr 2016, Steve Edwards wrote:


On Wed, 13 Apr 2016, Jeremy Kister wrote:

is there a way i can use the asterisk cli (or some other asterisky method) 
to recreate that extensions.conf ?


Will 'dialplan save' help?


I just tried this one. It writes the dialplan, but without the application 
arguements. Worthless.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] recreating extensions.conf from live dialplan ?

[Steve Edwards]

On Wed, 13 Apr 2016, Jeremy Kister wrote:

is there a way i can use the asterisk cli (or some other asterisky 
method) to recreate that extensions.conf ?


Will 'dialplan save' help?

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

--
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[asterisk-users] recreating extensions.conf from live dialplan ?

[Jeremy Kister]

with the slip of a finger, i destroyed by extensions.conf (grep -i > 
extensions.conf)


I have a backup that is dozens of hours of code old.

is there a way i can use the asterisk cli (or some other asterisky 
method) to recreate that extensions.conf ?



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[asterisk-users] E1 Extensions.conf

[Khaled W Chehab]

Hi,

I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card
5.0V (rev 02)) 4 ports

I want to make a  loop test between digium card E1  to test the
configuration of dahdi

What I want to do scenario is 

I connect port 1 and port4 in the digium card with E1 cable 

SIPcall--E1 Digium port 1---(Loop)E1 port 2sip extension local.

 

kindly can any can help me to draw this dialpan in the extensions.conf

 

 

Description  Alarms  IRQbpviol CRC4   Fra
Codi Options  LBO

T4XXP (PCI) Card 0 Span 1OK  0  0  0  CCS
HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)

T4XXP (PCI) Card 0 Span 2RED 0  0  0  CCS
HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)

T4XXP (PCI) Card 0 Span 3RED 0  0  0  CCS
HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)

T4XXP (PCI) Card 0 Span 4OK  0  0  0  CCS
HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)

 

Khaled  Chehab

   NGN Eng.

 

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 Operations Office - Lebanon

 Office : +961 1 868686 ext 115

 Mobile: +961 3 045212

 E-mail:   mailto:bs...@mg-tel.com kche...@xplorium.com

 MSN ID :khalidche...@hotmail.com  

 Web Site:  http://www.Xplorium.com http://www.Xplorium.com

 

 



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Re: [asterisk-users] E1 Extensions.conf

[Khaled W Chehab]

Find my dahdi config files below 

 

dahdi-channels.conf

 

; Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4
ClockSource

group=0,11

context=default

switchtype = euroisdn

signalling = pri_cpe

channel = 1-15,17-31

context = default

group = 63

 

; Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS/CRC4 RED

group=0,12

context=from-pstn

switchtype = euroisdn

signalling = pri_cpe

channel = 32-46,48-62

context = default

group = 63

 

; Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 HDB3/CCS/CRC4 RED

group=0,13

context=from-pstn

switchtype = euroisdn

signalling = pri_cpe

channel = 63-77,79-93

context = default

group = 63

 

; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 HDB3/CCS/CRC4

group=0,14

context=from-pstn

switchtype = euroisdn

signalling = pri_cpe

channel = 94-108,110-124

context = default

group = 63

 

Chan_dahdi.conf

[trunkgroups]

[channels]

language=en

context=default

signalling = pri_cpe

callwaiting=yes

hidecallerid=no

usecallingpres=yes

callwaitingcallerid=yes

threewaycalling=yes

transfer=yes

canpark=yes

cancallforward=yes

callreturn=yes

echocancel=no

echocancelwhenbridged=no

relaxdtmf=yes

usedistinctiveringdetection=yes

usecallingpres=yes

busydetect=yes

callprogress=yes

rxgain=2.0

txgain=2.0

#include dahdi-channels.conf

 

/etc/dahdi/system.conf

# Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4
ClockSource

span=1,1,0,ccs,hdb3,crc4

# termtype: te

bchan=1-15,17-31

dchan=16

echocanceller=mg2,1-15,17-31

 

# Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS/CRC4 RED

span=2,2,0,ccs,hdb3,crc4

# termtype: te

bchan=32-46,48-62

dchan=47

echocanceller=mg2,32-46,48-62

 

# Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 HDB3/CCS/CRC4 RED

span=3,3,0,ccs,hdb3,crc4

# termtype: te

bchan=63-77,79-93

dchan=78

echocanceller=mg2,63-77,79-93

 

# Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 HDB3/CCS/CRC4

span=4,4,0,ccs,hdb3,crc4

# termtype: te

bchan=94-108,110-124

dchan=109

echocanceller=mg2,94-108,110-124

 

# Global data

 

loadzone= us

defaultzone = us

 

Hi,

I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card
5.0V (rev 02)) 4 ports

I want to make a  loop test between digium card E1  to test the
configuration of dahdi

What I want to do scenario is 

I connect port 1 and port4 in the digium card with E1 cable 

SIPcall--E1 Digium port 1---(Loop)E1 port 2sip extension local.

 

kindly can any can help me to draw this dialpan in the extensions.conf

 

 

Description  Alarms  IRQbpviol CRC4   Fra
Codi Options  LBO

T4XXP (PCI) Card 0 Span 1OK  0  0  0  CCS
HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)

T4XXP (PCI) Card 0 Span 2RED 0  0  0  CCS
HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)

T4XXP (PCI) Card 0 Span 3RED 0  0  0  CCS
HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)

T4XXP (PCI) Card 0 Span 4OK  0  0  0  CCS
HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)

 

Khaled  Chehab

   NGN Eng.

 

 Untitled

 Operations Office - Lebanon

 Office : +961 1 868686 ext 115

 Mobile: +961 3 045212

 E-mail:  kche...@xplorium.com mailto:bs...@mg-tel.com 

 MSN ID :khalidche...@hotmail.com  

 Web Site: http://www.Xplorium.com

 

 

 

  _  

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Re: [asterisk-users] E1 Extensions.conf

[Tzafrir Cohen]

Hi,

On Mon, Nov 09, 2009 at 12:52:15PM +0200, Khaled W Chehab wrote:
 Hi,
 
 I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card
 5.0V (rev 02)) 4 ports
 
 I want to make a  loop test between digium card E1  to test the
 configuration of dahdi

This is fairly simple. But I figure it is best that you actually
understand what happens here.

 
 What I want to do scenario is 
 
 I connect port 1 and port4 in the digium card with E1 cable 
 
 SIPcall--E1 Digium port 1---(Loop)E1 port 2sip extension local.

In this scenario we have several different Asterisk channels:

1. SIPcall - E1-port1

Incoming call from SIP generates a SIP channel. The dialplan context for
it is set in sip.conf .

You want it to generate a call to some DAHDI channel. This could be done
using e.g. Dial(DAHDI/g1)  (why g1? To what channels does it refer? See
documentation in chan_dahdi.conf to see  why I set it like that. Much of
it is arbitrary).

2. E1-port1 - E1-port2

Loopback cable. 

It seems that you connected port 1 to port 4 rather than to port 2,
right?

3. E1-port2 - sip extension

Now we have an incoming DAHDI call. The dialplan context is set from
'context' in chan_dahdi.conf (where exactly?) . Now you'll probably need
to use some dialplan such as:

  Dial(SIP/your-local-sip-extension)

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] E1 Extensions.conf

[Khaled W Chehab]

Hi,

I have a digium card (digium, Inc. Wildcard TE405P quad-span T1/E1/J1 card
5.0V (rev 02)) 4 ports

I want to make a loop test between spans  on digium card in order to test
the spans.

 

I connect port 1 and port4 with cross E1 cable 

I am trying to do this scenario 

SIPcall-- Digium span 1---(Loop)Span 4sip
mailto:extens...@xx.xx.xx.xx extens...@xx.xx.xx.xx.

 

Kindly can you help me on how to forward the call from Span1-àSpan4 and then
from span4-à...@xx.xx.xx

 

My dahdi_channels.conf

; Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER)

group=1

context=default

switchtype = euroisdn

signalling = pri_net

channel = 1-15,17-31

context = default

;group = 63

 

; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4

group=4

;context=default

switchtype = euroisdn

signalling = pri_cpe

channel = 94-108,110-124

context = incomingck

;group = 63

-extensions.conf-

[default]

exten = _X.,1,Dial(DAHDI/G1/${EXTEN})

 

[incomingck]

exten = _X.,1,Dial(SIP/96123...@212.98.141.217,60)

 

Regards

 

 



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[asterisk-users] Static extensions.conf|ael, best practice, hints (was: Re: What do you use? .conf or AEL?)

[Philipp Kempgen]

Tilghman Lesher schrieb:
 On Wednesday 11 February 2009 13:39:58 Philipp Kempgen wrote:
 Tilghman Lesher schrieb:

  On Wed, 11 Feb 2009, Tilghman Lesher wrote:
   My viewpoint is that you should work on separation of your application
   code versus data, so that other than new development, your dialplan
   should be completely static and never need changing (other than, like
   I said, new development).

  I'd have a range of extensions, when dialled, it goes to the database,
  retrieves the list of channels, and dials those channels.  The web
  frontend would look exactly the same, but the data would go directly into
  a database, not taking an extra step to go into a dialplan, then reload
  the text file.

 How do you define the hints (for BLF, directed pickup, group
 pickup etc.)?
 
 As of 1.6, you can have pattern-match hints that query a database for the
 actual set of channels.

Two additional questions:

What about permissions that could be expressed by contexts without
actually using contexts (static dialplan, not to be modified by
a web interface)?

What about permissions that are more complicated than what can be
expressed by contexts?


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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Re: [asterisk-users] Static extensions.conf|ael, best practice, hints (was: Re: What do you use? .conf or AEL?)

[Tilghman Lesher]

On Monday 23 February 2009 04:49:15 Philipp Kempgen wrote:
 Tilghman Lesher schrieb:
  On Wednesday 11 February 2009 13:39:58 Philipp Kempgen wrote:
  Tilghman Lesher schrieb:
   On Wed, 11 Feb 2009, Tilghman Lesher wrote:
My viewpoint is that you should work on separation of your
application code versus data, so that other than new development,
your dialplan should be completely static and never need changing
(other than, like I said, new development).
  
   I'd have a range of extensions, when dialled, it goes to the database,
   retrieves the list of channels, and dials those channels.  The web
   frontend would look exactly the same, but the data would go directly
   into a database, not taking an extra step to go into a dialplan, then
   reload the text file.
 
  How do you define the hints (for BLF, directed pickup, group
  pickup etc.)?
 
  As of 1.6, you can have pattern-match hints that query a database for the
  actual set of channels.

 Two additional questions:

 What about permissions that could be expressed by contexts without
 actually using contexts (static dialplan, not to be modified by
 a web interface)?

 What about permissions that are more complicated than what can be
 expressed by contexts?

You've lost me.  I have no idea what you're talking about.

-- 
Tilghman

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[asterisk-users] Re: extensions.conf #include behaviour

[Tony Mountifield]

In article [EMAIL PROTECTED],
Chris Bagnall [EMAIL PROTECTED] wrote:
 
 A quick question regarding extensions.conf #include behaviour if I may. I'm 
 sure someone
 will know the answer off the top of their head...
 
 How does asterisk handle overloading of contexts. For example, say an 
 extension exists in
 extensions.conf as follows:
 
 [incoming]
 some stuff
 
 Then one includes a, b and c.conf, each of which also contains:
 
 [incoming]
 more stuff - but none identical to the other incoming sections
 
 Would the last #include overwrite the whole [incoming] context, or would it 
 simply append
 the new directives to it?

If Asterisk accepts it without error, then you can see how it has interpreted
it by using the CLI command show dialplan.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] Newbie extensions.conf question

[Bruce Reeves]

Chris,

Here is how I might use this, I have a context called inside, is where each
of my extensions is dialed from. On my home box it looks like this.

[inside]
exten = 1000,1,Dial(SIP/1000,20,t)

What I would probably do is add the Notify command to each of my extensions
before my Dial, like so

[inside]
exten = 1000,1,Notify(${CALLERIDNUM}|${CALLERIDNAME}|${EXTEN}/sunnybook)
exten = 1000,n,Dial(SIP/1000,20,t)

As a side note, if you are trying to get a screen pop, there are several
programs that connect to the manager API and will create a screen pop when
the a dial event is triggered. I mainly us snap, which runs on windows, and
I can configure it to watch a certain extension and display the call
information for that extension. So instead of having a special line in my
dial plan, I have a program filtering through events on the manage
interface. Hope this helps.

On 3/2/07, Chris Griffin [EMAIL PROTECTED] wrote:


I'm still stuck on just exactly where in my extensions.conf file I
should put the code below.


Chris Griffin
[EMAIL PROTECTED]



On Feb 28, 2007, at 9:55 PM, Patrick wrote:

On Wed, 2007-02-28 at 23:28 -0600, voiplist wrote:
 Thanks for the link..

 As for Google, I know how to use it. I searched for Sven Slezak's
 Notify and other variations and got Squat..

Yes I had that too initially. The trick is to remove the 's from Slezak.
Then the first link that pops up is the link I gave below.

 On 2/28/07, Patrick [EMAIL PROTECTED] wrote:
 On Wed, 2007-02-28 at 22:04 -0600, voiplist wrote:
 What does this module do?

 On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote:
 I've installed Sven Slezak's Notify module.

 http://mezzo.net/asterisk/app_notify.html

 Google is your friend.

 Regards,
 Patrick

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Bruce
Nortex Networks
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Re: [asterisk-users] Newbie extensions.conf question

[Chris Griffin]

I'm still stuck on just exactly where in my extensions.conf file I  
should put the code below. I'm running 1.2.14 of asterisk.


Chris Griffin
[EMAIL PROTECTED]


On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote:

I've installed Sven Slezak's Notify module. He gives the follow as an
example line to put into extensions.conf

exten = s,1000,Notify(${CALLERIDNUM}|${CALLERIDNAME}|${EXTEN}/
sunnybook)

I understand what is going on with this line but I don't know where
in the extensions.conf file to put it?

Thanks,
Chris Griffin
[EMAIL PROTECTED]



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Re: [asterisk-users] Newbie extensions.conf question

[Chris Griffin]

I'm still stuck on just exactly where in my extensions.conf file I  
should put the code below.



Chris Griffin
[EMAIL PROTECTED]



On Feb 28, 2007, at 9:55 PM, Patrick wrote:

On Wed, 2007-02-28 at 23:28 -0600, voiplist wrote:

Thanks for the link..

As for Google, I know how to use it. I searched for Sven Slezak's
Notify and other variations and got Squat..


Yes I had that too initially. The trick is to remove the 's from Slezak.
Then the first link that pops up is the link I gave below.


On 2/28/07, Patrick [EMAIL PROTECTED] wrote:

On Wed, 2007-02-28 at 22:04 -0600, voiplist wrote:

What does this module do?

On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote:

I've installed Sven Slezak's Notify module.


http://mezzo.net/asterisk/app_notify.html

Google is your friend.

Regards,
Patrick

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[asterisk-users] Newbie extensions.conf question

[Chris Griffin]

I've installed Sven Slezak's Notify module. He gives the follow as an  
example line to put into extensions.conf


exten = s,1000,Notify(${CALLERIDNUM}|${CALLERIDNAME}|${EXTEN}/ 
sunnybook)


I understand what is going on with this line but I don't know where  
in the extensions.conf file to put it?


Thanks,
Chris Griffin
[EMAIL PROTECTED]



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Re: [asterisk-users] Newbie extensions.conf question

[Mike Lynchfield]

try putting near the exten = 1000,1,dial stuff

On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote:


I've installed Sven Slezak's Notify module. He gives the follow as an
example line to put into extensions.conf

exten = s,1000,Notify(${CALLERIDNUM}|${CALLERIDNAME}|${EXTEN}/
sunnybook)

I understand what is going on with this line but I don't know where
in the extensions.conf file to put it?

Thanks,
Chris Griffin
[EMAIL PROTECTED]



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--
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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Re: [asterisk-users] Newbie extensions.conf question

[Chris Griffin]

There is nothing like that in my extensions.conf file. Maybe I should  
have mentioned I'm running 1.2.14.



Chris Griffin
[EMAIL PROTECTED]



On Feb 28, 2007, at 3:07 PM, Mike Lynchfield wrote:

try putting near the exten = 1000,1,dial stuff

On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote:
I've installed Sven Slezak's Notify module. He gives the follow as an
example line to put into extensions.conf

exten = s,1000,Notify(${CALLERIDNUM}|${CALLERIDNAME}|${EXTEN}/
sunnybook)

I understand what is going on with this line but I don't know where
in the extensions.conf file to put it?

Thanks,
Chris Griffin
[EMAIL PROTECTED]



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--
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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Re: [asterisk-users] Newbie extensions.conf question

[voiplist]

What does this module do?

On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote:

I've installed Sven Slezak's Notify module. He gives the follow as an
example line to put into extensions.conf

exten = s,1000,Notify(${CALLERIDNUM}|${CALLERIDNAME}|${EXTEN}/
sunnybook)

I understand what is going on with this line but I don't know where
in the extensions.conf file to put it?

Thanks,
Chris Griffin
[EMAIL PROTECTED]



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Re: [asterisk-users] Newbie extensions.conf question

[Patrick]

On Wed, 2007-02-28 at 22:04 -0600, voiplist wrote:
 What does this module do?
 
 On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote:
  I've installed Sven Slezak's Notify module. 

http://mezzo.net/asterisk/app_notify.html

Google is your friend.

Regards,
Patrick

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Re: [asterisk-users] Newbie extensions.conf question

[voiplist]

Thanks for the link..

As for Google, I know how to use it. I searched for Sven Slezak's
Notify and other variations and got Squat..


On 2/28/07, Patrick [EMAIL PROTECTED] wrote:

On Wed, 2007-02-28 at 22:04 -0600, voiplist wrote:
 What does this module do?

 On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote:
  I've installed Sven Slezak's Notify module.

http://mezzo.net/asterisk/app_notify.html

Google is your friend.

Regards,
Patrick

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Re: [asterisk-users] Newbie extensions.conf question

[Patrick]

On Wed, 2007-02-28 at 23:28 -0600, voiplist wrote:
 Thanks for the link..
 
 As for Google, I know how to use it. I searched for Sven Slezak's
 Notify and other variations and got Squat..

Yes I had that too initially. The trick is to remove the 's from Slezak.
Then the first link that pops up is the link I gave below.

 On 2/28/07, Patrick [EMAIL PROTECTED] wrote:
  On Wed, 2007-02-28 at 22:04 -0600, voiplist wrote:
   What does this module do?
  
   On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote:
I've installed Sven Slezak's Notify module.
 
  http://mezzo.net/asterisk/app_notify.html
 
  Google is your friend.
 
  Regards,
  Patrick
 
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Re: [asterisk-users] Re: extensions.conf strangeness

[Brian Candler]

On Thu, Oct 05, 2006 at 04:07:14PM +0200, Michael Neuhauser wrote:
 I've created and attached a one line patch (for 1.4 branch, r44464) that
 should give you the info you need (sort of). But be aware that I haven't
 tested it on 1.4 (only on 1.2, but things are different there). Only use
 this patch on a test system as it will generate massive amounts of
 output and will considerably slow down call handling.

Thank you. I could have written the printf() myself, I just wouldn't have
known where to put it :-)

I have applied it to trunk (r44544) and it generates output.

Unfortunately (or perhaps fortunately), now I'm running on trunk the problem
has gone away. That is, with my dialplan of

[internal]
include = extensions
include = outbound
include = invalid
include = test

[from-sip]
include = extensions
include = outbound
include = invalid
include = test

then both SIP phones and Zap phones work identically: dialling 611 gives
I'm sorry, that's not a valid extension, presumably because 'invalid' is
before 'test' (where 'invalid' matches _X!, and 'test' matches 611)

So I can only guess this is a 1.2 issue which has been fixed in trunk - or
else there was some uninitialised variable and the problem is now hidden.

Many thanks,

Brian.
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Re: [asterisk-users] Re: extensions.conf strangeness

[Brian Candler]

On Wed, Oct 04, 2006 at 12:20:40AM -0700, Martin Joseph wrote:
 Are there any debug tools which can show the thought process as a
 dial-plan is processed - for example, what patterns are tried and in what
 order?
 
 You can say show dialplan from the command line...
 
 Don't know if this helps?

Well, it shows each context as a separate list of tests, which at least
gives me the sort order. But that still doesn't explain why context A which
includes W,X,Y,Z behaves differently from context B which also includes
W,X,Y,Z

Is there a debug mode which can say:

dialplan: trying to match 611 against pattern _1X: failed
 dialplan: trying to match 611 against pattern _2X: failed
 dialplan: trying to match 611 against pattern _6X.: matched

?

Cheers,

Brian.
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Re: [asterisk-users] Re: extensions.conf strangeness

[Michael Neuhauser]

On Thu, 2006-10-05 at 11:12 +0100, Brian Candler wrote:
 Is there a debug mode which can say:
 
 dialplan: trying to match 611 against pattern _1X: failed
  dialplan: trying to match 611 against pattern _2X: failed
  dialplan: trying to match 611 against pattern _6X.: matched

No, there isn't (I assume to keep this central part as fast as possible,
i.e., even if (option_debug) ... costs time and pollutes the cache).

I've created and attached a one line patch (for 1.4 branch, r44464) that
should give you the info you need (sort of). But be aware that I haven't
tested it on 1.4 (only on 1.2, but things are different there). Only use
this patch on a test system as it will generate massive amounts of
output and will considerably slow down call handling.
-- 
Dr. Michael Neuhauser  mailto:[EMAIL PROTECTED]
Firmix Software GmbH  sip:[EMAIL PROTECTED]
Vienna/Austria/Europe   tel:+43-1-7890849-30
Linux Development and Services http://www.firmix.at/
Index: main/pbx.c
===
--- main/pbx.c	(revision 44464)
+++ main/pbx.c	(working copy)
@@ -952,6 +952,7 @@
 	while ( (eroot = ast_walk_context_extensions(tmp, eroot)) ) {
 		int match = extension_match_core(eroot-exten, exten, action);
 		/* 0 on fail, 1 on match, 2 on earlymatch */
+ast_log(LOG_NOTICE, [%s] match(%s, %s, %x) - %d\n, tmp-name, eroot-exten, exten, action, match);
 
 		if (!match || (eroot-matchcid  !matchcid(eroot-cidmatch, callerid)))
 			continue;	/* keep trying */
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[asterisk-users] Re: extensions.conf strangeness

[Martin Joseph]

On 2006-10-02 13:55:15 -0700, Brian Candler [EMAIL PROTECTED] said:


On Sun, Oct 01, 2006 at 03:55:45PM +0100, Marco Mouta wrote:


[invalid]

exten = _X!,1,Answer()
exten = _X!,2,Background(pbx-invalid)

Are you sure that your invalid context is correctly written?
I've never heard about this pattern match _X!
As far as i know the wild card is the .
So your invalid context should be:
[invalid]
exten = _X.,1,Answer()
exten = _X.,2,Background(pbx-invalid)
This may be the cause


_X! means match the pattern as soon as it possibly could. If you use _X.
then a timeout has to take place to see whether some other pattern might
match.

But your explanation still doesn't go into why it works differently in one
context than another. I guess I'm going to have to assume that Asterisk
dialplans are non-deterministic :-(

Are there any debug tools which can show the thought process as a
dial-plan is processed - for example, what patterns are tried and in what
order?


You can say show dialplan from the command line...

Don't know if this helps?

Marty


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[asterisk-users] sip.conf, extensions.conf

[ashok kumar]

 
Hi to all,
  I am just trying for Asterisk on my fedora core3 machine. here's my sip extensions config files


sip.conf

[general]
bindport=5060
bindaddr=0.0.0.0
allow=all
context=ECPT
localnet=192.168.0.1
localmask=255.255.255.0


[phone1]
type=friend
host=192.168.0.53
context=Embedded
callerid=ashok449


[phone2]
type=friend
host=192.168.0.22
context=Embedded
callerid=pramod450


[phone3]
type=friend
host=192.168.0.54
context=Embedded
callerid=pramod451


extensions.conf

[general]
static=yes
writeprotect=yes

[globals]
CONSOLE=Console/dsp
 
[local]
include=default
include=Embedded
include=ECPT


[ECPT]
exten=-.,1,congestion


[Embedded]
exten=449,1,dial(SIP/phone1,20)
exten=449,2,voicemail,u449
exten=450,1,dial(SIP/phone2,20)
exten=450,2,voicemail,u450

exten =451,1,dial(SIP/phone3,20)
exten =451,2,voicemail,u451

I need you people help for following things

1) how do i configure sip.conf to register my clients when host=dynamic?

2) how do i configure extensions.conf to make a call to PSTN ?

3) suggest a Softphone for asterisk on LAN.

4) how to establish a connection between two Asterisk servers?

thanks in advance,





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[Asterisk-Users] RE: extensions.conf

[andrutto]

Hi

Write a perl script that generates a mock 45,000 extensions.conf file, with 
45,000 incrementing extensions, throw in a couple of contexts. Start Asterisk 
and see what happens.

I will do that first thing in the morning :)

Actually i've done 50,000+ line dialplans using my Asterisk::LCR
dialplan generator, and asterisk has been just fine with it.

Sound of relief :)

Thanks for help I will let you know about the results.

Cheers


--
Zobacz nowosci salonu moto w Madrycie  http://link.interia.pl/f1961

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RE: [Asterisk-Users] newbiew extensions.conf question

[Tarpo, Louie]

We do extension by extension is our dialing plan because we have a wildcard at 
the end trapping all unused extensions and playing a this extension is not in 
use message and forwarding users into our IVR.  It depends on individual 
circumstances which works better.  We have 300 DIDs for our sip phones, and 
only 50 in use.  Those 50 are also not sequential extensions.  So it's less 
painful to approach this way for our circumstance.  If you had all of your 
extensions in use, the wildcard would be easier and cleaner.  Then if you 
needed to remove one, include a [not-in-service] context above the in use 
extensions.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
Walker
Sent: Thursday, August 04, 2005 9:58 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] newbiew extensions.conf question


But why do it that way?

Wouldn't:

exten = _72X,1,Dial(SIP/${EXTEN},50)

Be ideal? Or at least an easier way to expand the dialplan without mucho
administration?

Just a question...

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tarpo, Louie
Sent: Thursday, August 04, 2005 6:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] newbiew extensions.conf question

He means that exten = 720,1,macro(sipexten,${EXTEN}) is the template.

You'd write out the rest of the config file like so
exten = 720,1,macro(sipexten,${EXTEN})
exten = 721,1,macro(sipexten,${EXTEN})
exten = 722,1,macro(sipexten,${EXTEN})

and so forth.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
Walker
Sent: Thursday, August 04, 2005 6:22 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] newbiew extensions.conf question


That would make all callers have to call 720 as there is not other extension
defined. As a result, all calls would go to 720. ${EXTEN} would always be
720.

I don't follow your logic.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jj
Sent: Thursday, August 04, 2005 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] newbiew extensions.conf question

Right track, but it can be simplified even more

exten = 720,1,macro(sipexten,${EXTEN})


On Aug 4, 2005, at 5:25 PM, Tarpo, Louie wrote:

 We handled it by creating a macro which dials the exten, then sends  
 the call to voicemail.

 You could create it where each extension is handled seperately
 exten = 720,1,Macro(sipexten,720)
 exten = 721,1,Macro(sipexten,720)
 etc

 or you could handle them all in a group with wildcards
 exten = _72x,1,Macro(sipexten,${EXTEN})

 then the macro would look something like
 [macro-sipexten]
 exten = s,1,NoOp(${CallerIDNum})
 exten = s,2,Dial(SIP/${ARG1},24)
 exten = s,3,Goto(s-${DIALSTATUS}, 1)

 exten = s-NOANSWER,1,VoiceMail(u${ARG1});Send to  
 voicemail, play unavailable message
 exten = s-NOANSWER,2,Hangup

 exten = s-BUSY,1,VoiceMail(b${ARG1});Send to  
 voicemail, play busy message
 exten = s-BUSY,2,Hangup

 exten = _s-.,1,Goto(s-NOANSWER,1)

 Depends on your needs which way would work better.  We define  
 extension by extension individually, then have a wildcard at the  
 end that plays a message that says the extension is not in use and  
 then puts them in our main menu.  In case we have to remove or  
 change an extension individually.

 Louie


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Kenny  
 Kant
 Sent: Thursday, August 04, 2005 4:07 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] newbiew extensions.conf question


 I am newbie trying to setup about 12 Polycom Ip500's
 on an asterisk server.  I am working on my
 extensions.conf and am trying to make it so that all
 my extensions can dial each other. My extensions are
 number 720, 721, 722, 723 ..etc

 in my from-sip context I began doing entries such as:


 exten = 720,1,Dial(SIP/720,20)
 exten = 720,2,Voicemail(u720)


 exten = 721,1,Dial(SIP/721,20)
 exten = 721,2,Voicemail(u721)


 ..etc ..etc

 This is not a big deal for such a small number of
 extensions but I was thinking about larger installs..
 this would begin to suck.  Is there anyway around
 this?

 Thanks!

 Kenny




 
 Start your day with Yahoo! - make it your home page
 http://www.yahoo.com/r/hs

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Re: [Asterisk-Users] newbiew extensions.conf question

[jj]

On Aug 5, 2005, at 11:20 AM, Tarpo, Louie wrote:

We do extension by extension is our dialing plan because we have a  
wildcard at the end trapping all unused extensions and playing a  
this extension is not in use message and forwarding users into  
our IVR.  It depends on individual circumstances which works  
better.  We have 300 DIDs for our sip phones, and only 50 in use.   
Those 50 are also not sequential extensions.  So it's less painful  
to approach this way for our circumstance.  If you had all of your  
extensions in use, the wildcard would be easier and cleaner.  Then  
if you needed to remove one, include a [not-in-service] context  
above the in use extensions.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
Walker
Sent: Thursday, August 04, 2005 9:58 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] newbiew extensions.conf question


But why do it that way?

Wouldn't:

exten = _72X,1,Dial(SIP/${EXTEN},50)

Be ideal? Or at least an easier way to expand the dialplan without  
mucho

administration?

Just a question...


If all calls are handled exactly the same way then yes. But in my  
world all extensions are not the same.


As an example some have voicemail, others do not. some are sip some  
are zap.


By creating macros you have a macro for each class of extension and  
your dialplan calls appropriately, but you need a specified line for  
each. although a _match might catch all unspecified extensions would  
have to try it. I find it much easier to troubleshoot/read/support by  
more people to have each step explicitly spelled out. Although we do  
use exten matching on outdials - don't really want to enter every  
possible telephone number:)




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of  
Tarpo, Louie

Sent: Thursday, August 04, 2005 6:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] newbiew extensions.conf question

He means that exten = 720,1,macro(sipexten,${EXTEN}) is the template.

You'd write out the rest of the config file like so
exten = 720,1,macro(sipexten,${EXTEN})
exten = 721,1,macro(sipexten,${EXTEN})
exten = 722,1,macro(sipexten,${EXTEN})

and so forth.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
Walker
Sent: Thursday, August 04, 2005 6:22 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] newbiew extensions.conf question


That would make all callers have to call 720 as there is not other  
extension
defined. As a result, all calls would go to 720. ${EXTEN} would  
always be

720.

I don't follow your logic.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jj
Sent: Thursday, August 04, 2005 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] newbiew extensions.conf question

Right track, but it can be simplified even more

exten = 720,1,macro(sipexten,${EXTEN})


On Aug 4, 2005, at 5:25 PM, Tarpo, Louie wrote:



We handled it by creating a macro which dials the exten, then sends
the call to voicemail.

You could create it where each extension is handled seperately
exten = 720,1,Macro(sipexten,720)
exten = 721,1,Macro(sipexten,720)
etc

or you could handle them all in a group with wildcards
exten = _72x,1,Macro(sipexten,${EXTEN})

then the macro would look something like
[macro-sipexten]
exten = s,1,NoOp(${CallerIDNum})
exten = s,2,Dial(SIP/${ARG1},24)
exten = s,3,Goto(s-${DIALSTATUS}, 1)

exten = s-NOANSWER,1,VoiceMail(u${ARG1});Send to
voicemail, play unavailable message
exten = s-NOANSWER,2,Hangup

exten = s-BUSY,1,VoiceMail(b${ARG1});Send to
voicemail, play busy message
exten = s-BUSY,2,Hangup

exten = _s-.,1,Goto(s-NOANSWER,1)

Depends on your needs which way would work better.  We define
extension by extension individually, then have a wildcard at the
end that plays a message that says the extension is not in use and
then puts them in our main menu.  In case we have to remove or
change an extension individually.

Louie


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kenny
Kant
Sent: Thursday, August 04, 2005 4:07 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] newbiew extensions.conf question


I am newbie trying to setup about 12 Polycom Ip500's
on an asterisk server.  I am working on my
extensions.conf and am trying to make it so that all
my extensions can dial each other. My extensions are
number 720, 721, 722, 723 ..etc

in my from-sip context I began doing entries such as:


exten = 720,1,Dial(SIP/720,20)
exten = 720,2,Voicemail(u720)


exten = 721,1,Dial(SIP/721,20)
exten = 721,2,Voicemail(u721)


..etc ..etc

This is not a big deal for such a small number of
extensions but I was thinking about larger

RE: [Asterisk-Users] newbiew extensions.conf question

[Tarpo, Louie]

They both work.  Without a wildcard in that, you don't really gain anything by 
doing ${EXTEN} in that case except for changing the one number during 
copy/paste operations.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of jj
Sent: Thursday, August 04, 2005 5:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] newbiew extensions.conf question


Right track, but it can be simplified even more

exten = 720,1,macro(sipexten,${EXTEN})


On Aug 4, 2005, at 5:25 PM, Tarpo, Louie wrote:

 We handled it by creating a macro which dials the exten, then sends  
 the call to voicemail.

 You could create it where each extension is handled seperately
 exten = 720,1,Macro(sipexten,720)
 exten = 721,1,Macro(sipexten,720)
 etc

 or you could handle them all in a group with wildcards
 exten = _72x,1,Macro(sipexten,${EXTEN})

 then the macro would look something like
 [macro-sipexten]
 exten = s,1,NoOp(${CallerIDNum})
 exten = s,2,Dial(SIP/${ARG1},24)
 exten = s,3,Goto(s-${DIALSTATUS}, 1)

 exten = s-NOANSWER,1,VoiceMail(u${ARG1});Send to  
 voicemail, play unavailable message
 exten = s-NOANSWER,2,Hangup

 exten = s-BUSY,1,VoiceMail(b${ARG1});Send to  
 voicemail, play busy message
 exten = s-BUSY,2,Hangup

 exten = _s-.,1,Goto(s-NOANSWER,1)

 Depends on your needs which way would work better.  We define  
 extension by extension individually, then have a wildcard at the  
 end that plays a message that says the extension is not in use and  
 then puts them in our main menu.  In case we have to remove or  
 change an extension individually.

 Louie


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Kenny  
 Kant
 Sent: Thursday, August 04, 2005 4:07 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] newbiew extensions.conf question


 I am newbie trying to setup about 12 Polycom Ip500's
 on an asterisk server.  I am working on my
 extensions.conf and am trying to make it so that all
 my extensions can dial each other. My extensions are
 number 720, 721, 722, 723 ..etc

 in my from-sip context I began doing entries such as:


 exten = 720,1,Dial(SIP/720,20)
 exten = 720,2,Voicemail(u720)


 exten = 721,1,Dial(SIP/721,20)
 exten = 721,2,Voicemail(u721)


 ..etc ..etc

 This is not a big deal for such a small number of
 extensions but I was thinking about larger installs..
 this would begin to suck.  Is there anyway around
 this?

 Thanks!

 Kenny




 
 Start your day with Yahoo! - make it your home page
 http://www.yahoo.com/r/hs

 ___
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 http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] newbiew extensions.conf question

[Kenny Kant]

I am newbie trying to setup about 12 Polycom Ip500's
on an asterisk server.  I am working on my
extensions.conf and am trying to make it so that all
my extensions can dial each other. My extensions are
number 720, 721, 722, 723 ..etc 

in my from-sip context I began doing entries such as:


exten = 720,1,Dial(SIP/720,20)
exten = 720,2,Voicemail(u720)


exten = 721,1,Dial(SIP/721,20)
exten = 721,2,Voicemail(u721)


..etc ..etc

This is not a big deal for such a small number of
extensions but I was thinking about larger installs..
this would begin to suck.  Is there anyway around
this?

Thanks!

Kenny





Start your day with Yahoo! - make it your home page 
http://www.yahoo.com/r/hs 
 
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RE: [Asterisk-Users] newbiew extensions.conf question

[Jason Walker]

If all of your extensions are in the same schema (i.e. 7## or 7###) you
could do this:

Exten = _7XX,1,Dial(DEVICE/${EXTEN})
Exten = _7XX,2,Voicemail(u${EXTEN})

This would allow for any 7## number to call into the extension. ${EXTEN} is
the variable for the extension dialed. I am using DEVICE in case you
decide to use other methods or protocols - IAX/2, Zap, etc.

Hope that helps.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kenny Kant
Sent: Thursday, August 04, 2005 3:07 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] newbiew extensions.conf question

I am newbie trying to setup about 12 Polycom Ip500's
on an asterisk server.  I am working on my
extensions.conf and am trying to make it so that all
my extensions can dial each other. My extensions are
number 720, 721, 722, 723 ..etc 

in my from-sip context I began doing entries such as:


exten = 720,1,Dial(SIP/720,20)
exten = 720,2,Voicemail(u720)


exten = 721,1,Dial(SIP/721,20)
exten = 721,2,Voicemail(u721)


..etc ..etc

This is not a big deal for such a small number of
extensions but I was thinking about larger installs..
this would begin to suck.  Is there anyway around
this?

Thanks!

Kenny





Start your day with Yahoo! - make it your home page 
http://www.yahoo.com/r/hs 
 
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RE: [Asterisk-Users] newbiew extensions.conf question

[Tarpo, Louie]

We handled it by creating a macro which dials the exten, then sends the call to 
voicemail.  

You could create it where each extension is handled seperately
exten = 720,1,Macro(sipexten,720)
exten = 721,1,Macro(sipexten,720)
etc

or you could handle them all in a group with wildcards
exten = _72x,1,Macro(sipexten,${EXTEN})

then the macro would look something like 
[macro-sipexten]
exten = s,1,NoOp(${CallerIDNum})
exten = s,2,Dial(SIP/${ARG1},24)
exten = s,3,Goto(s-${DIALSTATUS}, 1)

exten = s-NOANSWER,1,VoiceMail(u${ARG1})   ;Send 
to voicemail, play unavailable message
exten = s-NOANSWER,2,Hangup

exten = s-BUSY,1,VoiceMail(b${ARG1})   ;Send to 
voicemail, play busy message
exten = s-BUSY,2,Hangup

exten = _s-.,1,Goto(s-NOANSWER,1)

Depends on your needs which way would work better.  We define extension by 
extension individually, then have a wildcard at the end that plays a message 
that says the extension is not in use and then puts them in our main menu.  In 
case we have to remove or change an extension individually.

Louie


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kenny Kant
Sent: Thursday, August 04, 2005 4:07 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] newbiew extensions.conf question


I am newbie trying to setup about 12 Polycom Ip500's
on an asterisk server.  I am working on my
extensions.conf and am trying to make it so that all
my extensions can dial each other. My extensions are
number 720, 721, 722, 723 ..etc 

in my from-sip context I began doing entries such as:


exten = 720,1,Dial(SIP/720,20)
exten = 720,2,Voicemail(u720)


exten = 721,1,Dial(SIP/721,20)
exten = 721,2,Voicemail(u721)


..etc ..etc

This is not a big deal for such a small number of
extensions but I was thinking about larger installs..
this would begin to suck.  Is there anyway around
this?

Thanks!

Kenny





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Re: [Asterisk-Users] newbiew extensions.conf question

[jj]

Right track, but it can be simplified even more

exten = 720,1,macro(sipexten,${EXTEN})


On Aug 4, 2005, at 5:25 PM, Tarpo, Louie wrote:

We handled it by creating a macro which dials the exten, then sends  
the call to voicemail.


You could create it where each extension is handled seperately
exten = 720,1,Macro(sipexten,720)
exten = 721,1,Macro(sipexten,720)
etc

or you could handle them all in a group with wildcards
exten = _72x,1,Macro(sipexten,${EXTEN})

then the macro would look something like
[macro-sipexten]
exten = s,1,NoOp(${CallerIDNum})
exten = s,2,Dial(SIP/${ARG1},24)
exten = s,3,Goto(s-${DIALSTATUS}, 1)

exten = s-NOANSWER,1,VoiceMail(u${ARG1});Send to  
voicemail, play unavailable message

exten = s-NOANSWER,2,Hangup

exten = s-BUSY,1,VoiceMail(b${ARG1});Send to  
voicemail, play busy message

exten = s-BUSY,2,Hangup

exten = _s-.,1,Goto(s-NOANSWER,1)

Depends on your needs which way would work better.  We define  
extension by extension individually, then have a wildcard at the  
end that plays a message that says the extension is not in use and  
then puts them in our main menu.  In case we have to remove or  
change an extension individually.


Louie


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kenny  
Kant

Sent: Thursday, August 04, 2005 4:07 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] newbiew extensions.conf question


I am newbie trying to setup about 12 Polycom Ip500's
on an asterisk server.  I am working on my
extensions.conf and am trying to make it so that all
my extensions can dial each other. My extensions are
number 720, 721, 722, 723 ..etc

in my from-sip context I began doing entries such as:


exten = 720,1,Dial(SIP/720,20)
exten = 720,2,Voicemail(u720)


exten = 721,1,Dial(SIP/721,20)
exten = 721,2,Voicemail(u721)


..etc ..etc

This is not a big deal for such a small number of
extensions but I was thinking about larger installs..
this would begin to suck.  Is there anyway around
this?

Thanks!

Kenny





Start your day with Yahoo! - make it your home page
http://www.yahoo.com/r/hs

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RE: [Asterisk-Users] newbiew extensions.conf question

[Jason Walker]

That would make all callers have to call 720 as there is not other extension
defined. As a result, all calls would go to 720. ${EXTEN} would always be
720.

I don't follow your logic.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jj
Sent: Thursday, August 04, 2005 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] newbiew extensions.conf question

Right track, but it can be simplified even more

exten = 720,1,macro(sipexten,${EXTEN})


On Aug 4, 2005, at 5:25 PM, Tarpo, Louie wrote:

 We handled it by creating a macro which dials the exten, then sends  
 the call to voicemail.

 You could create it where each extension is handled seperately
 exten = 720,1,Macro(sipexten,720)
 exten = 721,1,Macro(sipexten,720)
 etc

 or you could handle them all in a group with wildcards
 exten = _72x,1,Macro(sipexten,${EXTEN})

 then the macro would look something like
 [macro-sipexten]
 exten = s,1,NoOp(${CallerIDNum})
 exten = s,2,Dial(SIP/${ARG1},24)
 exten = s,3,Goto(s-${DIALSTATUS}, 1)

 exten = s-NOANSWER,1,VoiceMail(u${ARG1});Send to  
 voicemail, play unavailable message
 exten = s-NOANSWER,2,Hangup

 exten = s-BUSY,1,VoiceMail(b${ARG1});Send to  
 voicemail, play busy message
 exten = s-BUSY,2,Hangup

 exten = _s-.,1,Goto(s-NOANSWER,1)

 Depends on your needs which way would work better.  We define  
 extension by extension individually, then have a wildcard at the  
 end that plays a message that says the extension is not in use and  
 then puts them in our main menu.  In case we have to remove or  
 change an extension individually.

 Louie


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Kenny  
 Kant
 Sent: Thursday, August 04, 2005 4:07 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] newbiew extensions.conf question


 I am newbie trying to setup about 12 Polycom Ip500's
 on an asterisk server.  I am working on my
 extensions.conf and am trying to make it so that all
 my extensions can dial each other. My extensions are
 number 720, 721, 722, 723 ..etc

 in my from-sip context I began doing entries such as:


 exten = 720,1,Dial(SIP/720,20)
 exten = 720,2,Voicemail(u720)


 exten = 721,1,Dial(SIP/721,20)
 exten = 721,2,Voicemail(u721)


 ..etc ..etc

 This is not a big deal for such a small number of
 extensions but I was thinking about larger installs..
 this would begin to suck.  Is there anyway around
 this?

 Thanks!

 Kenny




 
 Start your day with Yahoo! - make it your home page
 http://www.yahoo.com/r/hs

 ___
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RE: [Asterisk-Users] newbiew extensions.conf question

[Tarpo, Louie]

He means that exten = 720,1,macro(sipexten,${EXTEN}) is the template.

You'd write out the rest of the config file like so
exten = 720,1,macro(sipexten,${EXTEN})
exten = 721,1,macro(sipexten,${EXTEN})
exten = 722,1,macro(sipexten,${EXTEN})

and so forth.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
Walker
Sent: Thursday, August 04, 2005 6:22 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] newbiew extensions.conf question


That would make all callers have to call 720 as there is not other extension
defined. As a result, all calls would go to 720. ${EXTEN} would always be
720.

I don't follow your logic.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jj
Sent: Thursday, August 04, 2005 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] newbiew extensions.conf question

Right track, but it can be simplified even more

exten = 720,1,macro(sipexten,${EXTEN})


On Aug 4, 2005, at 5:25 PM, Tarpo, Louie wrote:

 We handled it by creating a macro which dials the exten, then sends  
 the call to voicemail.

 You could create it where each extension is handled seperately
 exten = 720,1,Macro(sipexten,720)
 exten = 721,1,Macro(sipexten,720)
 etc

 or you could handle them all in a group with wildcards
 exten = _72x,1,Macro(sipexten,${EXTEN})

 then the macro would look something like
 [macro-sipexten]
 exten = s,1,NoOp(${CallerIDNum})
 exten = s,2,Dial(SIP/${ARG1},24)
 exten = s,3,Goto(s-${DIALSTATUS}, 1)

 exten = s-NOANSWER,1,VoiceMail(u${ARG1});Send to  
 voicemail, play unavailable message
 exten = s-NOANSWER,2,Hangup

 exten = s-BUSY,1,VoiceMail(b${ARG1});Send to  
 voicemail, play busy message
 exten = s-BUSY,2,Hangup

 exten = _s-.,1,Goto(s-NOANSWER,1)

 Depends on your needs which way would work better.  We define  
 extension by extension individually, then have a wildcard at the  
 end that plays a message that says the extension is not in use and  
 then puts them in our main menu.  In case we have to remove or  
 change an extension individually.

 Louie


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Kenny  
 Kant
 Sent: Thursday, August 04, 2005 4:07 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] newbiew extensions.conf question


 I am newbie trying to setup about 12 Polycom Ip500's
 on an asterisk server.  I am working on my
 extensions.conf and am trying to make it so that all
 my extensions can dial each other. My extensions are
 number 720, 721, 722, 723 ..etc

 in my from-sip context I began doing entries such as:


 exten = 720,1,Dial(SIP/720,20)
 exten = 720,2,Voicemail(u720)


 exten = 721,1,Dial(SIP/721,20)
 exten = 721,2,Voicemail(u721)


 ..etc ..etc

 This is not a big deal for such a small number of
 extensions but I was thinking about larger installs..
 this would begin to suck.  Is there anyway around
 this?

 Thanks!

 Kenny




 
 Start your day with Yahoo! - make it your home page
 http://www.yahoo.com/r/hs

 ___
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RE: [Asterisk-Users] newbiew extensions.conf question

[Jason Walker]

But why do it that way?

Wouldn't:

exten = _72X,1,Dial(SIP/${EXTEN},50)

Be ideal? Or at least an easier way to expand the dialplan without mucho
administration?

Just a question...

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tarpo, Louie
Sent: Thursday, August 04, 2005 6:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] newbiew extensions.conf question

He means that exten = 720,1,macro(sipexten,${EXTEN}) is the template.

You'd write out the rest of the config file like so
exten = 720,1,macro(sipexten,${EXTEN})
exten = 721,1,macro(sipexten,${EXTEN})
exten = 722,1,macro(sipexten,${EXTEN})

and so forth.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
Walker
Sent: Thursday, August 04, 2005 6:22 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] newbiew extensions.conf question


That would make all callers have to call 720 as there is not other extension
defined. As a result, all calls would go to 720. ${EXTEN} would always be
720.

I don't follow your logic.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jj
Sent: Thursday, August 04, 2005 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] newbiew extensions.conf question

Right track, but it can be simplified even more

exten = 720,1,macro(sipexten,${EXTEN})


On Aug 4, 2005, at 5:25 PM, Tarpo, Louie wrote:

 We handled it by creating a macro which dials the exten, then sends  
 the call to voicemail.

 You could create it where each extension is handled seperately
 exten = 720,1,Macro(sipexten,720)
 exten = 721,1,Macro(sipexten,720)
 etc

 or you could handle them all in a group with wildcards
 exten = _72x,1,Macro(sipexten,${EXTEN})

 then the macro would look something like
 [macro-sipexten]
 exten = s,1,NoOp(${CallerIDNum})
 exten = s,2,Dial(SIP/${ARG1},24)
 exten = s,3,Goto(s-${DIALSTATUS}, 1)

 exten = s-NOANSWER,1,VoiceMail(u${ARG1});Send to  
 voicemail, play unavailable message
 exten = s-NOANSWER,2,Hangup

 exten = s-BUSY,1,VoiceMail(b${ARG1});Send to  
 voicemail, play busy message
 exten = s-BUSY,2,Hangup

 exten = _s-.,1,Goto(s-NOANSWER,1)

 Depends on your needs which way would work better.  We define  
 extension by extension individually, then have a wildcard at the  
 end that plays a message that says the extension is not in use and  
 then puts them in our main menu.  In case we have to remove or  
 change an extension individually.

 Louie


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Kenny  
 Kant
 Sent: Thursday, August 04, 2005 4:07 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] newbiew extensions.conf question


 I am newbie trying to setup about 12 Polycom Ip500's
 on an asterisk server.  I am working on my
 extensions.conf and am trying to make it so that all
 my extensions can dial each other. My extensions are
 number 720, 721, 722, 723 ..etc

 in my from-sip context I began doing entries such as:


 exten = 720,1,Dial(SIP/720,20)
 exten = 720,2,Voicemail(u720)


 exten = 721,1,Dial(SIP/721,20)
 exten = 721,2,Voicemail(u721)


 ..etc ..etc

 This is not a big deal for such a small number of
 extensions but I was thinking about larger installs..
 this would begin to suck.  Is there anyway around
 this?

 Thanks!

 Kenny




 
 Start your day with Yahoo! - make it your home page
 http://www.yahoo.com/r/hs

 ___
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[Asterisk-Users] Convert extensions.conf INTO MySQL script

[Chris Coulthurst]

I swear I read somewhere on one of the MANY pages that there is a script out
there that can read the extensions.conf file and create the MySQL DB records
on the fly.  Anyone know where I look for such a thing?


Sure speeds up migration!

Chris Coulthurst
[EMAIL PROTECTED]
 


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Re: [Asterisk-Users] Convert extensions.conf INTO MySQL script

[Matthew Boehm]

Chris Coulthurst wrote:

I swear I read somewhere on one of the MANY pages that there is a script out
there that can read the extensions.conf file and create the MySQL DB records
on the fly.  Anyone know where I look for such a thing?


Sure speeds up migration!

Chris Coulthurst
[EMAIL PROTECTED]


That will be $10. Please pull up to the next window.

#!/usr/bin/perl -w

// originally downloaded from http://www.bkw.org/load.txt
//
// coding by pfn
//

use DBI;
use strict;
use POSIX;

if (@ARGV != 1) {
print STDERR Usage: load_res_config ast_config_file\n;
exit 1;
}
open(CONFIG_FILE, $ARGV[0]) || die $!;
my @lines;
my $cat_metric = -1; # incremented to 0 on first hit
my $var_metric = -1;
my $category;
while (CONFIG_FILE) {
my $line = $_;
chop($line);
my($var_name, $var_val);

next if ($line =~ /^\s*;/); # comment line skip

if ($line =~ /^\s*\[(.*?)\]/) {
$category = $1;
$var_metric = -1;
$cat_metric++;
} elsif ($line =~ /^\s*(\w+)\s*=\s*(.+)\s*;?.*$/ ||
   $line =~ /^\s*(\w+)\s*=\s*(.+)\s*;?.*$/) {
$var_metric++;
$var_name = $1;
$var_val  = $2;
} else {
next; # no match, skip
}

if ($var_metric = 0) {
my %hash = ('cat_metric' = $cat_metric,
'var_metric' = $var_metric,
'category'   = $category,
'var_name'   = $var_name,
'var_val'= $var_val);
push(@lines, \%hash);
}
}

close(CONFIG_FILE);

my $dbh;
$dbh = DBI-connect(dbi:mysql:dbname=asterisk, root, asdf) || die 
$DBI::errstr;

foreach my $row (@lines) {
print 
$row-{'cat_metric'}\t$row-{'category'}\t$row-{'var_metric'}\t$row-{'var_name'}\t$row-{'var_val'}\n;
my $sth = $dbh-prepare(INSERT into ast_config (filename, 
cat_metric, var_metric, category, var_name, var_val) values (?, ?, ?, ?, 
?, ?));

$sth-bind_param(1, $ARGV[0]);
$sth-bind_param(2, $row-{'cat_metric'});
$sth-bind_param(3, $row-{'var_metric'});
$sth-bind_param(4, $row-{'category'});
$sth-bind_param(5, $row-{'var_name'});
$sth-bind_param(6, $row-{'var_val'});
$sth-execute();
warn $sth-errstr if $sth-errstr;
}
$dbh-disconnect;

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[Asterisk-Users] Re: extensions.conf dial plan

[Tony Mountifield]

In article [EMAIL PROTECTED],
Georg P. Israel [EMAIL PROTECTED] wrote:
 Dear Asterisk users,
 
 I was wondering if anybody can tell me how to define a dial scheeme such
 that an incomming all first rings for e.g. 20 seconds on one set of
 phones and then after this time extends it's range onto a bigger set of
 phones.
 Basically, this is easy,
 
 I can do this in the extensions.con with 
 
 
 [ISDN-in]
 exten= 6201030,1,setcallerid(${CALLERID} ${CALLERID}|a)
 exten= 6201030,2,dial,${UserGroup1}|20|t
 exten= 6201030,3,dial,${UserGroup1UserGroup2}|60|t
 exten= 6201030,4,Voicemail2(u6201030)
 exten= 6201030,5,hangup
 exten= 6201030,302,Voicemail2(b6201030)
 
 
 But here is on major problem,
 
 in step 2, after 20 seconds, the call on the phones in Group1 will be
 terminated and then restarted in the bigger group (Group1Group2).
 The problem with this is, during the transition is a time gap of a view
 seconds on the phones from Group1. That means, if I lift up the head set
 during this gape, then I can loos the calls on those phones.
 
 Hence, I was wondering if I can set the dial proceadure such, that I
 have the calls for 80 seconds on the phone Group1, and after 20 seconds
 additionally on the phone Group2 without any interruption of the ringing
 on the other phones.

I don't have a proven answer, but here is an idea to try:

[ISDN-in]
exten= 6201030,1,SetCallerID(${CALLERID} ${CALLERID}|a)
exten= 6201030,2,Dial(${UserGroup1}Local/[EMAIL PROTECTED]|80|t)
exten= 6201030,3,Voicemail2(u6201030)
exten= 6201030,4,Hangup
exten= 6201030,302,Voicemail2(b6201030)

[ISDN-in-delayed]
exten= 6201030,1,Wait(20)
exten= 6201030,2,Dial(${UserGroup2}|60|t)

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] sip.conf extensions.conf

[Mauro Locatelli]

Title: Re: [Asterisk-Users] sip.conf extensions.conf







Hi, my sip.conf and my extensions.conf :)
I hope it's useful

**SIP.CONF**

[general]
port = 5060 ; port to bind for sip connections
bindaddr = 0.0.0.0 ; ip to bind for sip connections
context = default ; default context for incoming sip calls
externip = 222.99.99.22 ; Your external ip
localnet = 192.168.1.0/255.255.255.0 ;localnet and mask


disallow = all ; disallow all codecs, we want to enable,
allow=g726
allow=ulaw
allow=alaw
allow= gsm ; what we deem is necessary
allow= ilbc
allow= speex

register =
sipphonenumber:[EMAIL PROTECTED]/marlow-sip ;information
about sipphone

[proxy01.sipphone.com]
type=friend
username=sipphonenumber
secret=sipphonepwd
host=proxy01.sipphone.com
context=sipphone
nat=1


[marlow]
callerid=(marlow 3986)
username=marlow
type=friend
secret=marlowpwd
host=dynamic
context=internal
canreinvite=no
nat=1

[brandon]
callerid=(brandon 3986)
username=brandon
type=friend
secret=brandonpwd
host=dynamic
context=internal
canreinvite=no

[david]
callerid=(david 3988)
username=david
type=friend
secret=davidpwd
host=dynamic
context=internal
canreinvite=no
---

**EXTENSIONS.CONF**

[general]
static=yes
writeprotect=no

[globals]
MARLOW_CID=brandon
MARLOW_SIPPHONE=sipphonenumber
PHONE1=SIP/marlow ;unuseful for now it's only a try
PHONE2=SIP/brandon ;unuseful for now it's only a try
PHONE3=SIP/david ;unuseful for now it's only a try

[internal]
 include = from-sip
 include = sipphone
 include = tollfree
 include = 3986
 include = 3987
 include = 3988
 include = voicesystem

[voicesystem]

 exten = ,1,VoiceMailMain(${CALLERIDNUM}) ; extension  is the VM
system,go directly to callers VM
 exten = ,2,Hangup


[3986]
 exten = 3986,1,Dial(SIP/marlow,20) ; call SIP extension marlow
for 60 seconds,if extension 3986 is called
 exten = 3986,2,Voicemail(u3986) ; if we can't connect to
marlow or after seconds go to the unavail VM
 exten = 3986,102,Voicemail(b3986) ; if busy, go to the busy VM

[3987]
 exten = 3987,1,Dial(SIP/brandon,60) ; call SIP extension
brandon for 60 seconds,if extension 3986 is called
 exten = 3987,2,Voicemail(u3986) ; if we can't connect to
brandon or after seconds go to the unavail VM
 exten = 3987,102,Voicemail(b3986) ; if busy, go to the busy VM

[3988]
 exten = 3988,1,Dial(SIP/brandon,60) ; call SIP extension david
for 60 seconds,if extension 3986 is called
 exten = 3988,2,Voicemail(u3986) ; if we can't connect to
david or after seconds go to the unavail VM
 exten = 3988,102,Voicemail(b3986) ; if busy, go to the busy VM


[from-sip]
 ;
 ; default extension for calls from SIP
 ;
 ; calls from sipphone

 ;for receive call from sipphone and send it to local phone 3986 but don't
work:( and I don't know why
 exten = marlow-sip,1,SetCIDName(SIP - ${CALLERIDNAME}) ; indicate that the
call came through sipphone
 exten = marlow-sip,2,Dial(Local/[EMAIL PROTECTED]/n)


[outbound-internal]
 ;
 ; include local extensions
 ;
 include = internal

 ;
 ; include SIP accounts
 ;
 include = sipphone

 ;
 ; include tollfree calls
 ;
 ;include = tollfree

[default]
 ; include from-sip for default. We don't use it, but it might be a good idea
 include = from-sip
 include = sipphone
 include = internal

[sipphone]
; Official Sipphone example don't work very well
; exten = _1747.,1,Dial(SIP/[EMAIL PROTECTED]) ; set my
callerid and name
; exten = _1747.,2,Playback(notavail) ; this did
not work out
; exten = _1747.,3,Busy

;Approach to gateway guide
 exten = _1747.,1,SetCallerID(${MARLOW_CID} ${MARLOW_SIPPHONE}) ; set my
callerid and name
 exten =
_1747.,2,Dial,SIP/${EXTEN:[EMAIL PROTECTED] ; dial
the number i wish to dial
 exten = _1747.,3,Playback(invalid) ; this did
not work out
 exten = _1747.,4,Hangup
 exten = _1747.,103,Busy

[tollfree]
 ;
 ; terminate toll-free no.'s via fwdnet
 ;
;Use for call italian toll free
; +39 800
; exten = _39800.,1,SetCallerID(${MARLOW_SIPPHONE})
; exten = _39800.,1,Dial,SIP/[EMAIL PROTECTED]
; exten = _39800N.,1,Dial,Zap/1/${EXTEN:2}

; Use for call external PSTN number
 exten = _0X.,1,Dial,Zap/1/${EXTEN:1}
 exten = _0X.,2,Playback(invalid)
 exten = _0X.,3,Hangup
 exten = _0X.,103,Busy

;Use for call american Toll free
; +1-800
 exten = _1800.,1,SetCallerID(${MARLOW_SIPPHONE})
 exten = _1800.,2,Dial,SIP/[EMAIL PROTECTED]
 exten = _1800.,3,Playback(invalid)
 exten = _1800.,4,Hangup
 exten = _1800.,103,Busy
---







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[Asterisk-Users] sip.conf extensions.conf

[Mauro Locatelli]

I have an asterisk server(x100p wildcard) that function as a gateway.
I have some local soft phone (for example 3) and I want:
 
1- call from one internal softphone the other internal softphone
2- call out on the PSTN from internal softphone
3- call out on the sipphone.com
4- receive call from external PSTN and choose wich internal softphone ring
5- receive call from sipphone and choose wich internal softphone ring
 
I make sip.conf and extensions.conf but only 1,2,3 point work..
 
If someone is in my situation, and work all full, can send me his sip.conf and 
extensions.conf for compare?
 
Very very thanks and sorry for english..
Mauro
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Re: [Asterisk-Users] sip.conf extensions.conf

[Anand S. Katti]

Am too in same situation...but inmy case even 1,2,3 are not working. Can
you send you .conf files ?

thanks,
anand



On Fri, 5 Nov 2004, Mauro Locatelli wrote:

 I have an asterisk server(x100p wildcard) that function as a gateway.
 I have some local soft phone (for example 3) and I want:

 1- call from one internal softphone the other internal softphone
 2- call out on the PSTN from internal softphone
 3- call out on the sipphone.com
 4- receive call from external PSTN and choose wich internal softphone ring
 5- receive call from sipphone and choose wich internal softphone ring

 I make sip.conf and extensions.conf but only 1,2,3 point work..

 If someone is in my situation, and work all full, can send me his sip.conf and 
 extensions.conf for compare?

 Very very thanks and sorry for english..
 Mauro
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RE: [Asterisk-Users] sip.conf extensions.conf

[Fares Gianluca]

Hi Mauro, can u send me yours sip.conf and extension.conf ?

Ciao Gianluca

- Messaggio originale -
   Da: Mauro Locatelli[EMAIL PROTECTED]
   Inviato: 05/11/04 16.47.23
   A: Asterisk Users Mailing List - Non-Commercial Discussion[EMAIL PROTECTED]
   Oggetto: [Asterisk-Users] sip.conf extensions.conf
 I have an asterisk server(x100p wildcard) that function as a gateway.
   I have some local soft phone (for example 3) and I want:

   1- call from one internal softphone the other internal softphone
   2- call out on the PSTN from internal softphone
   3- call out on the sipphone.com
   4- receive call from external PSTN and choose wich internal softphone ring
   5- receive call from sipphone and choose wich internal softphone ring

   I make sip.conf and extensions.conf but only 1,2,3 point work..

   If someone is in my situation, and work all full, can send me his sip.conf and 
extensions.conf for compare?

   Very very thanks and sorry for english..
   Mauro
  

[Messaggio troncato. Toccare Modifica-Segna per il download per recuperare la 
restante parte.]

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R: [Asterisk-Users] sip.conf extensions.conf

[Mauro Locatelli]

Now I can't because the *.conf file are at work..
Monday I send all:)

Mauro

-Messaggio originale-
Da: [EMAIL PROTECTED] per conto di Anand S. Katti
Inviato: ven 05/11/2004 18.40
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [Asterisk-Users] sip.conf extensions.conf
 
Am too in same situation...but inmy case even 1,2,3 are not working. Can
you send you .conf files ?

thanks,
anand


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R: [Asterisk-Users] sip.conf extensions.conf

[Mauro Locatelli]

Ye monday morning I send all:)

Ciao Mauro

P.S.i have all at work..


-Messaggio originale-
Da: [EMAIL PROTECTED] per conto di Fares Gianluca
Inviato: sab 06/11/2004 0.07
A: [EMAIL PROTECTED]
Oggetto: RE: [Asterisk-Users] sip.conf extensions.conf
 
Hi Mauro, can u send me yours sip.conf and extension.conf ?

Ciao Gianluca


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Re: [Asterisk-Users] IVRs extensions.conf

[Hermann Wecke]

On Tue, 8 Jun 2004, Chris wrote:
 I'm trying to build an IVRs. anyone here can
 spare a sample extensions.conf? or maybe
 a link.

I found the example I think is one of the best to learn about IVR:
http://www.voip-info.org/wiki-Asterisk+Telemarketer+Torture
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[Asterisk-Users] IVRs extensions.conf

[Chris]

hi all,
 
I'm trying to build an IVRs. anyone here can
spare a sample extensions.conf? or maybe
a link.
 
 Thanks in advance!
 
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Re: [Asterisk-Users] IVRs extensions.conf

[Hermann Wecke]

On Tue, 8 Jun 2004, Chris wrote:
 I'm trying to build an IVRs. anyone here can
 spare a sample extensions.conf? or maybe
 a link.

http://www.voip-info.org/wiki-Asterisk+tips+ivr+menu
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf
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RE: [Asterisk-Users] Newbie extensions.conf I need to include [SMS] context.

[Gary Ruddock]

I have been up all night and I gotta go to bed.
If there's anyone out there using asterisk to send SMS text messages in the 
UK with BT please gis a clue. Do I need to get the latest asterisk CVS?


Could anyone be so kind as to tell me how to modify this dialplan to accept 
and send SMS text messages. Do I need to update my basic Asterisk to 
include SMS functionality? In the example contexts a reference is made to 
/usr/lib/asterisk/smsin and I can't find that file.

I know that [local] is executed first and it includes other contexts. I 
need to add these two contexts

[smsdial]   ; create and send a text message, expects number+message 
and
connect to 17094009
exten = _X.,1,SMS(${CALLERIDNUM},,${EXTEN},${CALLERIDNAME})
exten = _X.,2,SMS(${CALLERIDNUM})
exten = _X.,3,Hangup

and
[incoming]
exten = _XX/_8005875290,1,SMS(${EXTEN:3},a)
exten = _XX/_8005875290,2,System(/usr/lib/asterisk/smsin ${EXTEN:3})
exten = _XX/_80058752[0-8]0,1,SMS(${EXTEN:3}${CALLERIDNUM:8:1},a)
exten = _XX/_80058752[0-8]0,2,System(/usr/lib/asterisk/smsin 
${EXTEN:3}${CALLERIDNUM:8:1})
exten = _XX/_80058752X0,3,Hangup

***  my extensions.conf ***
[general]
static=yes
writeprotect=no
[globals]
TRUNK=Zap/g1; Trunk interface
TRUNKMSD=1  ; MSD digits to strip 
(usually 1 or 0)

[trunkint]
;exten = _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
;exten = _9011.,2,Congestion
[trunkld]
exten = _90XXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _90XXXNXX,2,Congestion
[trunklocal]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9NXX,2,Congestion
exten = _907N,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _907N,2,Congestion
[trunktollfree]
exten = _90800NX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _90800NX,2,Congestion
[international]
ignorepat = 9
include = longdistance
include = trunkint
[longdistance]
;ignorepat = 9
;include = local
include = trunkld
[local]
ignorepat = 9
;include = default
include = parkedcalls
include = trunklocal
include = trunktollfree
include = trunkld
exten = 6001,1,Dial(SIP/6001,20,tr)
exten = 6002,1,Dial(SIP/6002,20,tr)
exten = 07,1,Answer
exten = 07,2,wait(2)
exten = 07,3,playback(welcome)
exten = 07,4,GotoIf($[foo${CALLERIDNUM} = foo]?7:5)
exten = 
07,5,Queue(salesq,,https://swiftdrinks.com/admin/callerid.php?callerid=${CALLERIDNUM})
exten = 07,6,Hangup
exten = 07,7,Wait(2)
exten = 07,8,Playback(privacy-unident)
exten = 07,9,Hangup

exten = 2500,1,Dial(Zap/32,40)
exten = 2500,2,VoiceMail2(u2500)
exten = 2500,3,Hangup
exten = 2500,102,VoiceMail2(b2500)
exten = 2500,103,Hangup
exten = 2501,1,Dial(Zap/33,40)
exten = 2501,2,VoiceMail2(u2500)
exten = 2501,3,Hangup
exten = 2501,102,VoiceMail2(b2501)
exten = 2501,103,Hangup
exten = 81,1,AddQueueMember(salesq|Zap/32)
exten = 81,2,wait(1)
exten = 81,3,Playback(agent-loginok)
exten = 81,4,wait(1)
exten = 81,5,Hangup
exten = 82,1,RemoveQueueMember(salesq|Zap/32)
exten = 82,2,wait(1)
exten = 82,3,Playback(agent-loggedoff)
exten = 82,4,wait(1)
exten = 82,5,Hangup
exten = 95,3,Playback(agent-loginok)
exten = 95,4,wait(1)
exten = 95,5,Hangup
exten = 96,1,RemoveQueueMember(salesq|SIP/6001)
exten = 96,2,wait(1)
exten = 96,3,Playback(agent-loggedoff)
exten = 96,4,wait(1)
exten = 96,5,Hangup
exten = 97,1,AddQueueMember(salesq|SIP/6002)
exten = 97,2,wait(1)
exten = 97,3,Playback(agent-loginok)
exten = 97,4,wait(1)
exten = 97,5,Hangup
exten = 98,1,RemoveQueueMember(salesq|SIP/6002)
exten = 98,2,wait(1)
exten = 98,3,Playback(agent-loggedoff)
exten = 98,4,wait(1)
exten = 98,5,Hangup
[macro-stdexten]
exten = s,1,Dial(${ARG2},20)   ; Ring the 
interface, 20 seconds maximum
exten = s,2,Voicemail(u${ARG1}); If 
unavailable, send to voicemail w/ unavail announce
exten = s,3,Goto(default,s,1)  ; If they 
press #, return to start
exten = s,102,Voicemail(b${ARG1})  ; If busy, 
send to voicemail w/ busy announce
exten = s,103,Goto(default,s,1); If they 
press #, return to start

;[mainmenu]
;
; Example main menu context with submenu
;
;exten = s,1,Answer
;exten = s,2,Background(thanks); Thanks for calling press 
1 for sales, 2 for support, ...
;exten = 1,1,Goto(submenu,s,1)
;exten = 2,1,Hangup
;include = default
;
;[submenu]
;exten = s,1,Ringing   ; Make them 
comfortable with 2 seconds of ringback
;exten = s,2,Wait,2
;exten = s,3,Background(submenuopts)   ; Thanks for calling the sales 
department.  Press 1 for steve, 2 for...
;exten = 1,1,Goto(default,steve,1)
;exten = 2,1,Goto(default,mark,2)

[default]
;empty

I want to include a new context in my exensions.conf
I have read this page 
http://www.voip-info.org/wiki-Asterisk+howto+dial+plan and I can sort of 

RE: [Asterisk-Users] Newbie extensions.conf I need to include [SMS] context.

[Jay Milk]

Google on asterisk sms -- the first result links to a working example.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary Ruddock
Sent: Tuesday, May 25, 2004 1:22 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Newbie extensions.conf I need to include
[SMS] context.



I have been up all night and I gotta go to bed.

If there's anyone out there using asterisk to send SMS text messages in
the 
UK with BT please gis a clue. Do I need to get the latest asterisk CVS?


Could anyone be so kind as to tell me how to modify this dialplan to 
accept
and send SMS text messages. Do I need to update my basic Asterisk to 
include SMS functionality? In the example contexts a reference is made
to 
/usr/lib/asterisk/smsin and I can't find that file.


I know that [local] is executed first and it includes other contexts. I
need to add these two contexts

[smsdial]   ; create and send a text message, expects
number+message 
and
connect to 17094009
exten = _X.,1,SMS(${CALLERIDNUM},,${EXTEN},${CALLERIDNAME})
exten = _X.,2,SMS(${CALLERIDNUM})
exten = _X.,3,Hangup

and

[incoming]
exten = _XX/_8005875290,1,SMS(${EXTEN:3},a)
exten = _XX/_8005875290,2,System(/usr/lib/asterisk/smsin 
${EXTEN:3}) exten = 
_XX/_80058752[0-8]0,1,SMS(${EXTEN:3}${CALLERIDNUM:8:1},a)
exten = _XX/_80058752[0-8]0,2,System(/usr/lib/asterisk/smsin 
${EXTEN:3}${CALLERIDNUM:8:1})
exten = _XX/_80058752X0,3,Hangup


***  my extensions.conf ***

[general] static=yes
writeprotect=no

[globals]
TRUNK=Zap/g1; Trunk interface
TRUNKMSD=1  ; MSD digits to strip 
(usually 1 or 0)

[trunkint]
;exten = _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
;exten = _9011.,2,Congestion

[trunkld]
exten = _90XXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _90XXXNXX,2,Congestion

[trunklocal]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9NXX,2,Congestion

exten = _907N,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _907N,2,Congestion

[trunktollfree]
exten = _90800NX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _90800NX,2,Congestion

[international]
ignorepat = 9
include = longdistance
include = trunkint

[longdistance]
;ignorepat = 9
;include = local
include = trunkld

[local]
ignorepat = 9
;include = default
include = parkedcalls
include = trunklocal
include = trunktollfree
include = trunkld

exten = 6001,1,Dial(SIP/6001,20,tr)
exten = 6002,1,Dial(SIP/6002,20,tr)

exten = 07,1,Answer
exten = 07,2,wait(2)
exten = 07,3,playback(welcome)
exten = 07,4,GotoIf($[foo${CALLERIDNUM} = foo]?7:5)
exten =
07,5,Queue(salesq,,https://swiftdrinks.com/admin/callerid.php?calle
rid=${CALLERIDNUM})
exten = 07,6,Hangup
exten = 07,7,Wait(2)
exten = 07,8,Playback(privacy-unident)
exten = 07,9,Hangup

exten = 2500,1,Dial(Zap/32,40)
exten = 2500,2,VoiceMail2(u2500)
exten = 2500,3,Hangup
exten = 2500,102,VoiceMail2(b2500)
exten = 2500,103,Hangup

exten = 2501,1,Dial(Zap/33,40)
exten = 2501,2,VoiceMail2(u2500)
exten = 2501,3,Hangup
exten = 2501,102,VoiceMail2(b2501)
exten = 2501,103,Hangup

exten = 81,1,AddQueueMember(salesq|Zap/32)
exten = 81,2,wait(1)
exten = 81,3,Playback(agent-loginok)
exten = 81,4,wait(1)
exten = 81,5,Hangup

exten = 82,1,RemoveQueueMember(salesq|Zap/32)
exten = 82,2,wait(1)
exten = 82,3,Playback(agent-loggedoff)
exten = 82,4,wait(1)
exten = 82,5,Hangup

exten = 95,3,Playback(agent-loginok)
exten = 95,4,wait(1)
exten = 95,5,Hangup

exten = 96,1,RemoveQueueMember(salesq|SIP/6001)
exten = 96,2,wait(1)
exten = 96,3,Playback(agent-loggedoff)
exten = 96,4,wait(1)
exten = 96,5,Hangup

exten = 97,1,AddQueueMember(salesq|SIP/6002)
exten = 97,2,wait(1)
exten = 97,3,Playback(agent-loginok)
exten = 97,4,wait(1)
exten = 97,5,Hangup

exten = 98,1,RemoveQueueMember(salesq|SIP/6002)
exten = 98,2,wait(1)
exten = 98,3,Playback(agent-loggedoff)
exten = 98,4,wait(1)
exten = 98,5,Hangup

[macro-stdexten]
exten = s,1,Dial(${ARG2},20)   ; Ring
the 
interface, 20 seconds maximum
exten = s,2,Voicemail(u${ARG1}); If 
unavailable, send to voicemail w/ unavail announce
exten = s,3,Goto(default,s,1)  ; If
they 
press #, return to start
exten = s,102,Voicemail(b${ARG1})  ; If
busy, 
send to voicemail w/ busy announce
exten = s,103,Goto(default,s,1); If
they 
press #, return to start

;[mainmenu]
;
; Example main menu context with submenu
;
;exten = s,1,Answer
;exten = s,2,Background(thanks); Thanks for calling
press 
1 for sales, 2 for support, ...
;exten = 1,1,Goto(submenu,s,1)
;exten = 2,1,Hangup
;include = default
;
;[submenu]
;exten = s,1,Ringing   ; Make them 
comfortable with 2 seconds of ringback
;exten = s,2,Wait,2
;exten

[Asterisk-Users] Newbie extensions.conf I need to include [SMS] context.

[Gary Ruddock]

I want to include a new context in my exensions.conf
I have read this page http://www.voip-info.org/wiki-Asterisk+howto+dial+plan 
and I can sort of follow it?!

I have a context [local] that I know zapata.conf points to, I have edited 
extensions.conf and put in my phone, sip and iax extensions. I want to add 
an sms context.

I understand that all calls go through my [local] context and I have other 
contexts that get included into [local] for long distance and freefone 
numbers.

At a guess would I put the code below in extensions.conf and include 
[smsdial] into the [local] context? I have read a page on extensions.conf 
parsing, would I include [smsdial] at the end of [local]?

Please help, cos I have to do the same for [fax].
[smsdial]   ; create and send a text message, expects number+message and
connect to 17094009
exten = _X.,1,SMS(${CALLERIDNUM},,${EXTEN},${CALLERIDNAME})
exten = _X.,2,SMS(${CALLERIDNUM})
exten = _X.,3,Hangup
_
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RE: [Asterisk-Users] Newbie extensions.conf I need to include [SMS] context.

[Gary Ruddock]

Could anyone be so kind as to tell me how to modify this dialplan to accept 
and send SMS text messages. Do I need to update my basic Asterisk to include 
SMS functionality? In the example contexts a reference is made to 
/usr/lib/asterisk/smsin and I can't find that file.

I know that [local] is executed first and it includes other contexts. I need 
to add these two contexts

[smsdial]   ; create and send a text message, expects number+message and
connect to 17094009
exten = _X.,1,SMS(${CALLERIDNUM},,${EXTEN},${CALLERIDNAME})
exten = _X.,2,SMS(${CALLERIDNUM})
exten = _X.,3,Hangup
and
[incoming]
exten = _XX/_8005875290,1,SMS(${EXTEN:3},a)
exten = _XX/_8005875290,2,System(/usr/lib/asterisk/smsin ${EXTEN:3})
exten = _XX/_80058752[0-8]0,1,SMS(${EXTEN:3}${CALLERIDNUM:8:1},a)
exten = _XX/_80058752[0-8]0,2,System(/usr/lib/asterisk/smsin 
${EXTEN:3}${CALLERIDNUM:8:1})
exten = _XX/_80058752X0,3,Hangup

***  my extensions.conf ***
[general]
static=yes
writeprotect=no
[globals]
TRUNK=Zap/g1; Trunk interface
TRUNKMSD=1  ; MSD digits to strip 
(usually 1 or 0)

[trunkint]
;exten = _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
;exten = _9011.,2,Congestion
[trunkld]
exten = _90XXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _90XXXNXX,2,Congestion
[trunklocal]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9NXX,2,Congestion
exten = _907N,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _907N,2,Congestion
[trunktollfree]
exten = _90800NX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _90800NX,2,Congestion
[international]
ignorepat = 9
include = longdistance
include = trunkint
[longdistance]
;ignorepat = 9
;include = local
include = trunkld
[local]
ignorepat = 9
;include = default
include = parkedcalls
include = trunklocal
include = trunktollfree
include = trunkld
exten = 6001,1,Dial(SIP/6001,20,tr)
exten = 6002,1,Dial(SIP/6002,20,tr)
exten = 07,1,Answer
exten = 07,2,wait(2)
exten = 07,3,playback(welcome)
exten = 07,4,GotoIf($[foo${CALLERIDNUM} = foo]?7:5)
exten = 
07,5,Queue(salesq,,https://swiftdrinks.com/admin/callerid.php?callerid=${CALLERIDNUM})
exten = 07,6,Hangup
exten = 07,7,Wait(2)
exten = 07,8,Playback(privacy-unident)
exten = 07,9,Hangup

exten = 2500,1,Dial(Zap/32,40)
exten = 2500,2,VoiceMail2(u2500)
exten = 2500,3,Hangup
exten = 2500,102,VoiceMail2(b2500)
exten = 2500,103,Hangup
exten = 2501,1,Dial(Zap/33,40)
exten = 2501,2,VoiceMail2(u2500)
exten = 2501,3,Hangup
exten = 2501,102,VoiceMail2(b2501)
exten = 2501,103,Hangup
exten = 81,1,AddQueueMember(salesq|Zap/32)
exten = 81,2,wait(1)
exten = 81,3,Playback(agent-loginok)
exten = 81,4,wait(1)
exten = 81,5,Hangup
exten = 82,1,RemoveQueueMember(salesq|Zap/32)
exten = 82,2,wait(1)
exten = 82,3,Playback(agent-loggedoff)
exten = 82,4,wait(1)
exten = 82,5,Hangup
exten = 95,3,Playback(agent-loginok)
exten = 95,4,wait(1)
exten = 95,5,Hangup
exten = 96,1,RemoveQueueMember(salesq|SIP/6001)
exten = 96,2,wait(1)
exten = 96,3,Playback(agent-loggedoff)
exten = 96,4,wait(1)
exten = 96,5,Hangup
exten = 97,1,AddQueueMember(salesq|SIP/6002)
exten = 97,2,wait(1)
exten = 97,3,Playback(agent-loginok)
exten = 97,4,wait(1)
exten = 97,5,Hangup
exten = 98,1,RemoveQueueMember(salesq|SIP/6002)
exten = 98,2,wait(1)
exten = 98,3,Playback(agent-loggedoff)
exten = 98,4,wait(1)
exten = 98,5,Hangup
[macro-stdexten]
exten = s,1,Dial(${ARG2},20)   ; Ring the 
interface, 20 seconds maximum
exten = s,2,Voicemail(u${ARG1}); If 
unavailable, send to voicemail w/ unavail announce
exten = s,3,Goto(default,s,1)  ; If they 
press #, return to start
exten = s,102,Voicemail(b${ARG1})  ; If busy, 
send to voicemail w/ busy announce
exten = s,103,Goto(default,s,1); If they 
press #, return to start

;[mainmenu]
;
; Example main menu context with submenu
;
;exten = s,1,Answer
;exten = s,2,Background(thanks); Thanks for calling press 
1 for sales, 2 for support, ...
;exten = 1,1,Goto(submenu,s,1)
;exten = 2,1,Hangup
;include = default
;
;[submenu]
;exten = s,1,Ringing   ; Make them 
comfortable with 2 seconds of ringback
;exten = s,2,Wait,2
;exten = s,3,Background(submenuopts)   ; Thanks for calling the sales 
department.  Press 1 for steve, 2 for...
;exten = 1,1,Goto(default,steve,1)
;exten = 2,1,Goto(default,mark,2)

[default]
;empty

I want to include a new context in my exensions.conf
I have read this page 
http://www.voip-info.org/wiki-Asterisk+howto+dial+plan and I can sort of 
follow it?!

I have a context [local] that I know zapata.conf points to, I have edited 
extensions.conf and put in my phone, sip and iax extensions. I want to add 
an sms context.

I understand that all