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Thread: So many configuration files!
So many configuration files!
user name
 2006-07-11 19:21:03 

  


  

    
		first: download the latest version 1.2.5 had some bugs and is already several months old.

Depending on how you want your asterisk to behave will be the amount of files you'll need to mess with. Let's say you want a very basic installation with some SIP phones (hard or soft), then you'll have to deal with sip.conf and extensions.conf only.... so everything else is just vanity 

Usually you would like to have some voicemail, conference rooms, music on hold, pick up your neighbours extension, dial another asterisk or an IAX softphone, and PSTN access, then change some configs in voicemail.conf, meetme.conf, musiconhold.conf, features.conf, iax.conf and zapata.conf respectively.

Want more action?

Manage asterisk from an external application and mess with manager.conf,  change the way logs are being saved and CRMs with logger.conf and  the cdr_ *.conf files,  try some text to speach (TTS) with festival.conf

Feel like you are in the right track?

try dealing with any ".c" file, recompile asterisk and make it behave just the way you always dream of (btw if it works you might want to share your new feature with all of us  )



Alyed 
		

		
				
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I'm working with Asterisk 1.2.5 to get a working system.

There are 50 Asterisk configuration files in /etc/asterisk.
Are they _all_ called by Asterisk or are some only used in a #include?

Is there any way to get a list of which ones Asterisk uses by default?
There is only a single #include file and it doesn't even exist.

I have only messed with 4 files so far.
Are there any more I should be editing?
Which ones could be safely ignored?

So far the system is just SIP with Zaptel to be added next.

The 4 files I have changed are:
sip.conf
extensions.conf
extensions_additional.conf
voicemail.conf

My list of files in /etc/asterisk - sorted most recent last:
------------------------------------------------------------
lbalinda asterisk # ls -1tr
zapata.conf
vpb.conf
telcordia-1.adsi
skinny.conf
sip_notify.conf
rtp.conf
rpt.conf
res_odbc.conf
queues.conf
privacy.conf
phone.conf
oss.conf
osp.conf
musiconhold.conf
modules.conf
modem.conf
misdn.conf
mgcp.conf
meetme.conf
manager.conf
logger.conf
indications.conf
iaxprov.conf
iax.conf
festival.conf
features.conf
extensions.ael
extconfig.conf
enum.conf
dundi.conf
dnsmgr.conf
codecs.conf
cdr_tds.conf
cdr_pgsql.conf
cdr_odbc.conf
cdr_manager.conf
cdr_custom.conf
cdr.conf
asterisk.conf
asterisk.adsi
alsa.conf
alarmreceiver.conf
agents.conf
adtranvofr.conf
adsi.conf
sip.conf
extensions.conf
extensions_additional.conf
voicemail.conf


-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
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    So many configuration files!
  


  

     user name

     2006-07-17 15:27:23
  


  

    

  


  

    For what it's worth, any version over 1.2.6 doesn't work for me since they have URI parsing issues dealing with Sonus switches..
-Brett


On 7/11/06, Alyed Tzompa
 <simitel.com">alyed.tzompasimitel.com> wrote:



		first: download the latest version 1.2.5 had some bugs and is already several months old.

Depending on how you want your asterisk to behave will be the amount of files you'll need to mess with. Let's say you want a very basic installation with some SIP phones (hard or soft), then you'll have to deal with 
sip.conf and extensions.conf only.... so everything else is just vanity 

Usually you would like to have some voicemail, conference rooms, music on hold, pick up your neighbours extension, dial another asterisk or an IAX softphone, and PSTN access, then change some configs in 
voicemail.conf, meetme.conf, musiconhold.conf, features.conf, iax.conf and zapata.conf respectively.

Want more action?

Manage asterisk from an external application and mess with manager.conf,  change the way logs are being saved and CRMs with 
logger.conf and  the cdr_ *.conf files,&nbsp; try some text to speach (TTS) with festival.conf

Feel like you are in the right track?

try dealing with any ".c&quot; file, recompile asterisk and make it behave just the way you always dream of (btw if it works you might want to share your new feature with all of us  )




Alyed 
		

		
				
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I'm working with Asterisk 1.2.5 to get a working system.

There are 50 Asterisk configuration files in /etc/asterisk.
Are they _all_ called by Asterisk or are some only used in a #include?

Is there any way to get a list of which ones Asterisk uses by default?

There is only a single #include file and it doesn't even exist.

I have only messed with 4 files so far.
Are there any more I should be editing?
Which ones could be safely ignored?

So far the system is just SIP with Zaptel to be added next.


The 4 files I have changed are:
sip.conf
extensions.conf
extensions_additional.conf
voicemail.conf

My list of files in /etc/asterisk - sorted most recent last:
------------------------------------------------------------

lbalinda asterisk # ls -1tr
zapata.conf
vpb.conf
telcordia-1.adsi
skinny.conf
sip_notify.conf
rtp.conf
rpt.conf
res_odbc.conf
queues.conf
privacy.conf
phone.conf
oss.conf
osp.conf

musiconhold.conf
modules.conf
modem.conf
misdn.conf
mgcp.conf
meetme.conf
manager.conf
logger.conf
indications.conf
iaxprov.conf
iax.conf
festival.conf
features.conf
extensions.ael

extconfig.conf
enum.conf
dundi.conf
dnsmgr.conf
codecs.conf
cdr_tds.conf
cdr_pgsql.conf
cdr_odbc.conf
cdr_manager.conf
cdr_custom.conf
cdr.conf
asterisk.conf
asterisk.adsi
alsa.conf

alarmreceiver.conf
agents.conf
adtranvofr.conf
adsi.conf
sip.conf
extensions.conf
extensions_additional.conf
voicemail.conf


-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux

_______________________________________________
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Easynews.com --

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    Session Border Controllers
  


  

     user name

     2006-07-17 16:14:26
  


  

    
Anyone got any experience with SBC's?  We've recently
installed 2
Netrake nCite SE's and are pretty happy with them.  The
amount of
overhead for configuration and network management is pretty
high though
and I've got a colleague that would like other
recommendations.  We
looked at Juniper but trunking between SBC's just doesn't
work (I'm not
sure how an SBC is supposed work without proper trunking but
whatever).
I'd prefer a hardware solution that has dsp's for audio
work instead of
a software based SBC, but I'm not ruling them out.

Chris Tooley

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    So many configuration files!
  


  

     user name

     2006-07-17 16:09:36
  


  

    
On Mon, 2006-07-17 at 10:27 -0500, Brett Nemeroff wrote:
> For what it's worth, any version over 1.2.6 doesn't
work for me since
> they have URI parsing issues dealing with Sonus
switches..
> -Brett

We're currently using 1.2.6 with no problems.  We're
connected to Sonus
at one provider and Veraz at another.  I'm not sure exactly
which Sonus
load they're running but we've had no issues with them.

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    So many configuration files!
  


  

     user name

     2006-07-17 17:37:57
  


  

    
Chris,
1.2.6 works great with Sonus, but I think you'll find that
1.2.7 and
up won't work at all.. Primary this is because the Sonus
puts TEL URI
options into the front part of the SIP URI. This usually
happens when
there is call portability information in the call. A Sonus
SIP URI
will end up looking like:
sip:5128881234;rn=5128880099;npdi=yes10.1.2.3;dtg=gtovpninsi0001tx

Asterisk doesn't expect to get those URI options before the

sign. So
it chokes.. In fact, it'll think this call is for s5128881234 which
is all sorts of wrong.. Let me know how it works for you if
you test
it out.. BTW, this is in the digium bugtracker.. I still
need to test
and provide traces..
-Brett


On 7/17/06, Chris Tooley <ctooleyunwiredbuyer.com> wrote:
> On Mon, 2006-07-17 at 10:27 -0500, Brett Nemeroff
wrote:
> > For what it's worth, any version over 1.2.6
doesn't work for me since
> > they have URI parsing issues dealing with Sonus
switches..
> > -Brett
>
> We're currently using 1.2.6 with no problems.  We're
connected to Sonus
> at one provider and Veraz at another.  I'm not sure
exactly which Sonus
> load they're running but we've had no issues with
them.
>
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Groupbybent.com
> http://buzzard.onr.com/mailman/listinfo/austi
n-asterisk-users-group
> AAUG Web Site: http://aaug.bybent.com/
>
_______________________________________________
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Austin-Asterisk-Users-Groupbybent.com
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    So many configuration files!
  


  

     user name

     2006-07-17 17:58:29
  


  

    
We're currently running 1.2.7 in our office connected to
CommPartners
and I think we're connected to their Sonus box.  It worked
fine but we
only did limited testing.  Number portability probably
didn't play into
that at all.

On Mon, 2006-07-17 at 12:37 -0500, Brett Nemeroff wrote:
> Chris,
> 1.2.6 works great with Sonus, but I think you'll find
that 1.2.7 and
> up won't work at all.. Primary this is because the
Sonus puts TEL URI
> options into the front part of the SIP URI. This
usually happens when
> there is call portability information in the call. A
Sonus SIP URI
> will end up looking like:
> sip:5128881234;rn=5128880099;npdi=yes10.1.2.3;dtg=gtovpninsi0001tx
> 
> Asterisk doesn't expect to get those URI options
before the  sign. So
> it chokes.. In fact, it'll think this call is for
s5128881234 which
> is all sorts of wrong.. Let me know how it works for
you if you test
> it out.. BTW, this is in the digium bugtracker.. I
still need to test
> and provide traces..
> -Brett
> 
> 
> On 7/17/06, Chris Tooley <ctooleyunwiredbuyer.com> wrote:
> > On Mon, 2006-07-17 at 10:27 -0500, Brett Nemeroff
wrote:
> > > For what it's worth, any version over 1.2.6
doesn't work for me since
> > > they have URI parsing issues dealing with
Sonus switches..
> > > -Brett
> >
> > We're currently using 1.2.6 with no problems. 
We're connected to Sonus
> > at one provider and Veraz at another.  I'm not
sure exactly which Sonus
> > load they're running but we've had no issues
with them.
> >
> > _______________________________________________
> > Austin-Asterisk-Users-Group mailing list
> > Austin-Asterisk-Users-Groupbybent.com
> > http://buzzard.onr.com/mailman/listinfo/austi
n-asterisk-users-group
> > AAUG Web Site: http://aaug.bybent.com/
> >
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Groupbybent.com
> http://buzzard.onr.com/mailman/listinfo/austi
n-asterisk-users-group
> AAUG Web Site: http://aaug.bybent.com/
-- 
Chris Tooley
Unwired Buyer, Inc
o 512-646-1507
f 512-305-0704
m 512-659-2498

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