date:20061107

[asterisk-users] Re: Do my messages come through?

2006-11-07 Thread Martin Joseph


On 2006-11-06 13:16:50 -0800, Christian [EMAIL PROTECTED] said:


Hi all,
DO my messages come through to the list? I have had some problems wiht 
my  email client here.


Looks like your spell checker has issues also...



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[asterisk-users] Desired apps

2006-11-07 Thread Justin Newman

Is there a list of apps or desired features for users?


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[asterisk-users] Asterisk and FreeTDS 0.64 or 0.63

2006-11-07 Thread RR


Hello all,

just curious if anyone's successfully compiled (*) with the latest
FreeTDS code/driver. The Makefile in (*) seems to only take care of
0.63 or older. I tried to muck around with it a bit into tricking to
compile for not just 0.63 but anything later than 0.62 but it seems to
crap out complaining about CDR modules, which I really don't need.
It's been a while since I tried it but I seriously doubt there's any
dev. done to focus on intergarting MSSQL or non-open-source DBs with
(*). If someone's done it or knows how to do it or even can tell me if
it's even worth it, then I'd really appreciate your comments.

Best Regards,
\R
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Re: [asterisk-users] Fast detection of unreachable SIP clients?

2006-11-07 Thread Dmitry Ivanov

On Monday 06 November 2006 16:41, Matt wrote:
 This should work.. please make sure you have qualify=yes on in
 your sip.conf file for each of your sip entries.

Now it works. Thank you!


 On 11/6/06, Dmitry Ivanov [EMAIL PROTECTED] wrote:
  Hello!
 
  I have this in my dialplan:
 
  Dial(SIP/${ext}, 300);
  switch(${DIALSTATUS}) {
  case BUSY:
  Busy();
  break;
  default:
  Hangup();
  };
 
  This means that SIP phone will ring for max. five minutes if
  phone can be contacted. When SIP phone is turned off or
  there is no connectivity, calling party hears many many
  seconds of silence. I want Dial() to return CHANUNAVAIL if
  there was no SIP response from the phone within 1 or 2
  seconds. In this case, calling party will hear out of
  range message similar to mobile networks. Is this possible?

-- 
Dmitry Ivanov
Network engineer
Telecentrs Riga, Latvia
[EMAIL PROTECTED]
(+371) 7160235

Weather at Riga Intl (EVRA/RIX): Tuesday 07 November 2006 09:50,9 
km/h S,2°C,1003 hPa,,,  
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[asterisk-users] Snom 360 flickering screen

2006-11-07 Thread Nick Hoffman

Hi guys. I just bought and configured a Snom 360 and have noticed that the 
LCD is constantly flickering at a rate of ~10-15Hz (that's a guess). 
Either way, it's very distracting. Has anyone else encountered this 
before? Any solutions?

Cheers,
-- Nick
E: [EMAIL PROTECTED]
P: +61 7 5591 3588
F: +61 7 5591 6588

If you receive this email by mistake, please notify us and do not make any 
use of the email.  We do not waive any privilege, confidentiality or 
copyright associated with it.
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[asterisk-users] Asterisk Showing 404 not found when calling from third party SIP server (newbie question)

2006-11-07 Thread Alok Mohapatra









Hi All,


I have installed Asterisk Successfully and configure a out bound trunk for
another SIP server so that if Ill dial 777123 from an asterisk-registered-phone
then it will dial to the phone extension(123)-registered in the third party
server.



But my problem is that the reverse is not happening, that
is I am not able to call from Third party SIP server to Asterisk extensions.



Actually the third party SIP Server is sending request to
Asterisk for the extension registered but Asterisk sending the response 404 not
found where as the actually the extension really exist.



I have included the all extensions contexts with the
incoming contexts.

Is there anything to configure for incoming contexts. 



Is there any way to know which trunk is accepting the
incoming calls 





Thanks and Regards

Alok Mohapatra










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Re: [asterisk-users] Dial plan Question

2006-11-07 Thread Anselm Martin Hoffmeister

Am Dienstag, den 07.11.2006, 02:29 -0500 schrieb Doug Crompton:
 I am trying to do something that I see describe in a book and it is not
 working
 
 In my sip.conf, I have in my [fxo] context=from-pstn
 
 I then have in extensions.conf
 
 [from-pstn]
 
 exten s,1,answer()
 exten s,2,playback(blah)
 
 etc.
 
 It never answers but if I do this
 
 [from-pstn]
 
 exten _x.,1,answer()
 exten _x.,2,playback(blah)
 
 it works.  Why does the 's' extension not work here?

If fxo means your SIP provider, and you register with him, a specific
extension will be called. Which one shall be called can be selected by
the last parameter of the register statement, e.g.

register = 075741:[EMAIL PROTECTED]:5060/492281234567

will cause the incoming calls to appear in extension 492281234567.

Comes in handy if you have several accounts with a single SIP provider:
This way, you can simply distinguish the outward phone number for which
the call came in.

BR
Anselm

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[asterisk-users] Upgrading sox

2006-11-07 Thread René Christensen


Hi,

I'm currently running an * version 1.2.13 and sox version 12.17.5. I want to 
upgrade sox to the newest release ( 12.18.2 ); need mp3 support.

But how do I make the upgrade.
Do I need to recompile asterisk afterwards?
If I make a  sox -h after a reboot  I can see the new version is running 
but is that enough?


_
Log på MSN Messenger direkte på nettet  http://webmessenger.msn.com/

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[asterisk-users] Mapping CLI'S in Dialplan

2006-11-07 Thread Scott Pinhorne









Hi All



I am not sure what I wish to do it possible but I would like
to see if you guys know any better.



I have a site who has the extensions: 1231, 1232. 1233, 1234



Each of these users can dial each other on the extension
number an also has an external CLI mapped to them.

On all internal calls or calls to services such as call
forwarding their Caller ID is: Name 



What I would like to have happen is have the Caller ID
changed to the CLI only when they make an offnet call.

So what I am saying is I need to match an extension to a CLI
and reset the Caller ID.



Many Thanks

SP










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Re: [asterisk-users] Mapping CLI'S in Dialplan

2006-11-07 Thread Benjamin Jacob


Your offnet calls will be more than 4 digits, so use that to ur advantage.
so, for internal calls,
exten = _,1,Set(CALLERID(all)=Name ${CALLERID(num)})
or if u dont want to change the CLID at all.. dont do anything..
exten = _,1,NoOp(nothing)

else, for all external calls(4 digits)
exten = _X.,1,Set(CALLERID(num)=urDID)


cheerz
- Ben.

Scott Pinhorne wrote:


Hi All

 

I am not sure what I wish to do it possible but I would like to see if 
you guys know any better.


 


I have a site who has the extensions: 1231, 1232. 1233, 1234

 

Each of these users can dial each other on the extension number an 
also has an external CLI mapped to them.


On all internal calls or calls to services such as call forwarding 
their Caller ID is: Name 


 

What I would like to have happen is have the Caller ID changed to the 
CLI only when they make an offnet call.


So what I am saying is I need to match an extension to a CLI and reset 
the Caller ID.


 


Many Thanks

SP

 

 




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Re: [asterisk-users] Upgrading sox

2006-11-07 Thread Tzafrir Cohen

On Tue, Nov 07, 2006 at 11:14:43AM +0100, René Christensen wrote:
 Hi,
 
 I'm currently running an * version 1.2.13 and sox version 12.17.5. I want 
 to upgrade sox to the newest release ( 12.18.2 ); need mp3 support.
 But how do I make the upgrade.
 Do I need to recompile asterisk afterwards?

No. Asterisk is not linked with sox.

 If I make a  sox -h after a reboot  I can see the new version is running 
 but is that enough?

A reboot should not be needed either.

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

2006-11-07 Thread Szabó András

Hi!Then please just tell me if it's even possible, as i cannot find any configuration to allow unknown codecs to be used in reinvited calls
.My question is that is it possible or impossible to handle this with asterisk?Thanks!AndrásOn 11/5/06, Szabó András
[EMAIL PROTECTED] wrote:Hi!I want to tell asterisk to
simply pass-through any codecs that my phones support. I have to use
codecs that are not popular and implemented by a third-party, asterisk
has nothing to do with them.
I've made a test with g722 (that asterisk doesn't support), i've set all my two snom 300
phones to support only g722 and asterisk declined the sip invitation.
That is bad for me. Is it possible that asterisk asks the called phone
not the codecs that asterisk supports but the codecs that the calling
phone supports?
What i want:phone1-asterisk: hello, i'm calling phone2, codecs possible: g722asterisk-phone2: hello, you have a call from phone1, codecs possible: g722phone2-asterisk: ok, let it be g722
... chit chat ...But asterisk does this:phone1-asterisk: hello, i'm calling phone2, codecs possible: g722asterisk-phone2: hello, you have a call from phone1, codecs possible: alaw, ulaw etc. (but not g722)
phone2-asterisk: no way, media not supported! (cannot agree in a codec)asterisk-phone1: beep beep beepAny ideas? Is it possible? Is it not possible?I've searched over

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