Re: [asterisk-users] Dial Plan Issues

2008-09-29 Thread Steve Murphy 
On Mon, 2008-09-29 at 14:51 +, Tariq .. wrote:
> the following two lines exist in the extensions_additional.conf
> 
>  
> [from-max]
> exten => _X,1,Answer
> exten => _X,n,Queue(8000,tr,,)
>  
> and it DOES exist in the output of the 'show dialplan'
> 
>  [ Context 'from-max' created by 'pbx_config' ]
>   '_X' =>   1. Answer()
> [pbx_config]
> 2. Queue(8000|tr||)
> [pbx_config]
>  
> yet my system doesn't use it to route
>  
> regards
> 

Tariq--

Maybe I missed a message or something, but I don't see a response to 
Tzafrir's request to see /etc/asterisk/extensions.conf.
extensions_additional.conf is not extensions.conf; and unless
extensions.conf includes it, it will never be a part of your dialplan.

You did mention that you were using trixbox in your original question,
so we referred you to a trixbox mailing list, because rumors have it
that trixbox does complicated things in their dialplan to accomplish
their goals, and most folks in this mailing list (but not all) 
don't play much with trixbox. 

But if you are not using trixbox, then you might look in your 
extensions.conf to answer these questions. 

Another resource you have to investigate the dialplan is in the
CLI of asterisk; you can say "dialplan show", or "dialplan show
from-max"
to see if the from-max context has been included.

when the pbx_config module (module load pbx_config.so) loads, it
will read in /etc/asterisk/extensions.conf; if it is not there,
that module will not complete the loading process.

If want us to evaluate why your dialplan is not working, show us the
dialplan in extensions.conf.

murf




> 
> __________
> > Date: Sun, 28 Sep 2008 23:31:46 +0300
> > From: [EMAIL PROTECTED]
> > To: asterisk-users@lists.digium.com
> > Subject: Re: [asterisk-users] Dial Plan Issues
> > 
> > On Sun, Sep 28, 2008 at 08:13:08PM +, Tariq .. wrote:
> > > this is not a TrixBOX .. i'm asking a simply question.. why
> doesn't asterisk read the dial plan!! 
> > 
> > What is the dialplan?
> > 
> > ls -ld /etc/asterisk /etc/asterisk/extensions.conf
> > 
> > And what is the contents of extensions.conf ?
> > 
> > What is the output of 'dialplan show' from the CLI?
> > 
-- 
Steve Murphy
Software Developer
Digium


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Re: [asterisk-users] Dial Plan Issues

2008-09-29 Thread Tariq .. 
the following two lines exist in the extensions_additional.conf
 [from-max]exten => _X,1,Answerexten => _X,n,Queue(8000,tr,,)
 
and it DOES exist in the output of the 'show dialplan'
 [ Context 'from-max' created by 'pbx_config' ]  '_X' =>   1. Answer()  
 [pbx_config]2. 
Queue(8000|tr||)   [pbx_config]
 
yet my system doesn't use it to route
 
regards




AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308

> Date: Sun, 28 Sep 2008 23:31:46 +0300> From: [EMAIL PROTECTED]> To: 
> asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Dial Plan 
> Issues> > On Sun, Sep 28, 2008 at 08:13:08PM +, Tariq .. wrote:> > this 
> is not a TrixBOX .. i'm asking a simply question.. why doesn't asterisk read 
> the dial plan!! > > What is the dialplan?> > ls -ld /etc/asterisk 
> /etc/asterisk/extensions.conf> > And what is the contents of extensions.conf 
> ?> > What is the output of 'dialplan show' from the CLI?> > -- > Tzafrir 
> Cohen> icq#16849755 jabber:[EMAIL PROTECTED]> +972-50-7952406 mailto:[EMAIL 
> PROTECTED]> http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir> > 
> ___> -- Bandwidth and Colocation 
> Provided by http://www.api-digital.com --> > AstriCon 2008 - September 22 - 
> 25 Phoenix, Arizona> Register Now: http://www.astricon.net> > asterisk-users 
> mailing list> To UNSUBSCRIBE or update options visit:> 
> http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Dial Plan Issues

2008-09-28 Thread Tzafrir Cohen 
On Sun, Sep 28, 2008 at 08:13:08PM +, Tariq .. wrote:
> this is not a TrixBOX .. i'm asking a simply question.. why doesn't asterisk 
> read the dial plan!!  

What is the dialplan?

ls -ld /etc/asterisk /etc/asterisk/extensions.conf

And what is the contents of extensions.conf ?

What is the output of 'dialplan show' from the CLI?

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Dial Plan Issues

2008-09-28 Thread Tariq .. 
this is not a TrixBOX .. i'm asking a simply question.. why doesn't asterisk 
read the dial plan!!  


AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308

Date: Sun, 28 Sep 2008 10:00:56 -0400From: [EMAIL PROTECTED]: [EMAIL 
PROTECTED]: Re: [asterisk-users] Dial Plan Issues
This is a better question asked on a Fonality list.  Maybe they have a 
manual.Thanks,Steve Totaro
On Thu, Sep 25, 2008 at 10:21 AM, Tariq .. <[EMAIL PROTECTED]> wrote:
Greetings,i have two asterisk servers running on Centos with asterisk 1.4.21 
and trixbox..i tried to creat an SIP link between both servers and i discovered 
that one of my servers is not allowing the other to send calls while it is 
possible in the opposit direction..i have the same exact settings for the 
extensions.confi tried with another friend of mine.. and connected to his 
server.. and it didn't allow him to send me calls..so my question is..is it 
possible that my server is not accepting any context ? it only runs the ones 
that come default with Trixbix.. like chanspy, ext-local, from-trunk... and so 
on..what can i do to avoide this problem?? i can't rebuild a new box this one 
is a production server and i wasn't making tests.. i was connecting two of my 
employer's servers with each other..regardsAHD 
Tarek SawahIntegrated Digital SystemsCCNA, MCSE, RHCE, VoIPSyria: +963 944 
618286USA: +1 347 562 
2308_Want to do 
more with Windows Live? Learn "10 hidden secrets" from 
Jamie.http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!550F681DAD532637!5295.entry?ocid=TXT_TAGLM_WL_domore_092008___--
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Thanks,Steve Totaro1.888.777.18881.240.938.1212 (cell)
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Re: [asterisk-users] Dial Plan Issues

2008-09-28 Thread Steve Totaro 
This is a better question asked on a Fonality list.  Maybe they have a
manual.

Thanks,
Steve Totaro

On Thu, Sep 25, 2008 at 10:21 AM, Tariq .. <[EMAIL PROTECTED]> wrote:

>
> Greetings,
> i have two asterisk servers running on Centos with asterisk 1.4.21 and
> trixbox..
> i tried to creat an SIP link between both servers and i discovered that one
> of my servers is not allowing the other to send calls while it is possible
> in the opposit direction..
> i have the same exact settings for the extensions.conf
> i tried with another friend of mine.. and connected to his server.. and it
> didn't allow him to send me calls..
> so my question is..
> is it possible that my server is not accepting any context ? it only runs
> the ones that come default with Trixbix.. like chanspy, ext-local,
> from-trunk... and so on..
> what can i do to avoide this problem?? i can't rebuild a new box this one
> is a production server and i wasn't making tests.. i was connecting two of
> my employer's servers with each other..
> regards
> 
>
>
>
>
> AHD Tarek Sawah
>
>
> Integrated Digital Systems
>
>
> CCNA, MCSE, RHCE, VoIP
>
>
> Syria: +963 944 618286
>
>
> USA: +1 347 562 2308
>
>
>
> _
> Want to do more with Windows Live? Learn "10 hidden secrets" from Jamie.
>
> http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!550F681DAD532637!5295.entry?ocid=TXT_TAGLM_WL_domore_092008
> ___
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> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Thanks,
Steve Totaro
1.888.777.1888
1.240.938.1212 (cell)
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Re: [asterisk-users] Dial Plan Issues

2008-09-28 Thread Tariq .. 

no.. it's directly connected to the internet.. it's not an issue of accepting 
calls.. see.. the problem is the call gets to the server.. the server tries to 
route it.. 
but as if the dial plan is not there.. it rejects the call because it doesn't 
know what to do with it.. 
for example of my SIP.Conf
 
[5003]
type=peer
qualify=yes
port=5060
nat=yes
host=HOSTIP
allow=all
dial=SIP/5003
context=from-smarttech
canreinvite=no
call-limit=50
deny=0.0.0.0/0.0.0.0
permit=HOSTIP/255.255.255.255
 
Extensions.conf
[from-smarttech]
exten => fax,1,Goto(ext-fax,in_fax,1)
exten => s,n,Set(__FROM_DID=${EXTEN})
exten => s,n,Gosub(app-blacklist-check,s,1)
exten => s,n,GotoIf($[ "${CALLERID(name)}" != "" ] ?cidok)
exten => s,n,Set(CALLERID(name)=${CALLERID(num)})
exten => s,n(cidok),Noop(CallerID is ${CALLERID(all)})
exten => s,n,Set(__CALLINGPRES_SV=${CALLINGPRES_${CALLINGPRES}})
exten => s,n,SetCallerPres(allowed_not_screened)
exten => s,n,Goto(ext-queues,8004,1)
let's say smarttech is a voip provider.. which forwards calls to my user on 
their system .. now my server is supposed to route those calls according to the 
dial plan.. 
the same exact settings worked like magic on another server.. but on this 
server.. it just as if the context and the dial plan does not exist.!!!
any idea?




AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308

Date: Fri, 26 Sep 2008 11:55:45 -0500From: [EMAIL PROTECTED]: [EMAIL 
PROTECTED]: Re: [asterisk-users] Dial Plan Issues
Steve Murphy wrote: 
On Thu, 2008-09-25 at 14:21 +, Tariq .. wrote:
  
Greetings,
i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox..
i tried to creat an SIP link between both servers and i discovered that one of 
my servers is not allowing the other to send calls while it is possible in the 
opposit direction.. 
i have the same exact settings for the extensions.conf 
i tried with another friend of mine.. and connected to his server.. and it 
didn't allow him to send me calls.. 
so my question is.. 
is it possible that my server is not accepting any context ? it only runs the 
ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... 
and so on.. 
what can i do to avoide this problem?? i can't rebuild a new box this one is a 
production server and i wasn't making tests.. i was connecting two of my 
employer's servers with each other..
regards



Tariq--

You might try a trixbox users mailing list.
There might be a few trixbox users hanging around in 
this group who might be able to help, but your
chances are much better in that list.

murf

  The server that is not accepting calls is not behind a NAT firewall by any 
chance is it?
_
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Re: [asterisk-users] Dial Plan Issues

2008-09-26 Thread Brent Davidson 

Steve Murphy wrote:

On Thu, 2008-09-25 at 14:21 +, Tariq .. wrote:
  

Greetings,
i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox..
i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls while it is possible in the opposit direction.. 
i have the same exact settings for the extensions.conf 
i tried with another friend of mine.. and connected to his server.. and it didn't allow him to send me calls.. 
so my question is.. 
is it possible that my server is not accepting any context ? it only runs the ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... and so on.. 
what can i do to avoide this problem?? i can't rebuild a new box this one is a production server and i wasn't making tests.. i was connecting two of my employer's servers with each other..

regards





Tariq--

You might try a trixbox users mailing list.
There might be a few trixbox users hanging around in 
this group who might be able to help, but your

chances are much better in that list.

murf

  


The server that is not accepting calls is not behind a NAT firewall by 
any chance is it?
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Re: [asterisk-users] Dial Plan Issues

2008-09-25 Thread Steve Murphy 
On Thu, 2008-09-25 at 14:21 +, Tariq .. wrote:
> Greetings,
> i have two asterisk servers running on Centos with asterisk 1.4.21 and 
> trixbox..
> i tried to creat an SIP link between both servers and i discovered that one 
> of my servers is not allowing the other to send calls while it is possible in 
> the opposit direction.. 
> i have the same exact settings for the extensions.conf 
> i tried with another friend of mine.. and connected to his server.. and it 
> didn't allow him to send me calls.. 
> so my question is.. 
> is it possible that my server is not accepting any context ? it only runs the 
> ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... 
> and so on.. 
> what can i do to avoide this problem?? i can't rebuild a new box this one is 
> a production server and i wasn't making tests.. i was connecting two of my 
> employer's servers with each other..
> regards
> 
> 
> 
Tariq--

You might try a trixbox users mailing list.
There might be a few trixbox users hanging around in 
this group who might be able to help, but your
chances are much better in that list.

murf

> 
> 
> AHD Tarek Sawah
> 
> 
> Integrated Digital Systems
> 
> 
> CCNA, MCSE, RHCE, VoIP
> 
> 
> Syria: +963 944 618286
> 
> 
> USA: +1 347 562 2308
> 

-- 
Steve Murphy
Software Developer
Digium


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[asterisk-users] Dial Plan Issues

2008-09-25 Thread Tariq .. 

Greetings,
i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox..
i tried to creat an SIP link between both servers and i discovered that one of 
my servers is not allowing the other to send calls while it is possible in the 
opposit direction.. 
i have the same exact settings for the extensions.conf 
i tried with another friend of mine.. and connected to his server.. and it 
didn't allow him to send me calls.. 
so my question is.. 
is it possible that my server is not accepting any context ? it only runs the 
ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... 
and so on.. 
what can i do to avoide this problem?? i can't rebuild a new box this one is a 
production server and i wasn't making tests.. i was connecting two of my 
employer's servers with each other..
regards





AHD Tarek Sawah


Integrated Digital Systems


CCNA, MCSE, RHCE, VoIP


Syria: +963 944 618286


USA: +1 347 562 2308



_
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