No subject


Tue Sep 5 14:32:44 MST 2006 

fax tone.

It then tries to redirect it, and prints the following message :

Redirecting Zap/2-1 to fax extension

According to the source, it does this only if it matches a "fax"
extension in the current context.

I don't have a "fax" extension, but a wildcard one (_.). I would like
these detections to be simply ignored. Is there any way to do it ?

-- 
Nicolas Bougues
Axialys Interactive

--__--__--

Message: 7
Date: Fri, 12 Mar 2004 12:54:46 +0100
From: Alessio Focardi <afoc at interconnessioni.it>
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] call bridge
Reply-To: asterisk-users at lists.digium.com

Hi all,

I would like to have Asterisk bridge 2 calls with this schema

-inbound call comes in
-the caller id is passed to an external script
-the external script replies with a phone number
-an outbound call to the number provided by the script is made
-if the outgoing call is answered we have to bridge inbound/outbound
calls
-if there is no answer/busy call is diverted to a voicebox

what would you suggest to archive such goal ?

The purpose is to connect our customers to field technicians without
giving them
their mobile phone number ... I think that is a very common issue in
our market :)

Tnx for any help you can give me !

-- 
Best regards,
 Alessio                          mailto:afoc at interconnessioni.it



--__--__--

Message: 8
Date: Fri, 12 Mar 2004 09:10:08 -0300
From: Daniel Bichara <daniel at bichara.com.br>
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Native Bridge and Billing
Reply-To: asterisk-users at lists.digium.com

Hi all,

I am connecting two * (A and B) using a third * (C) as passthru and
billing control. All connections are IAX-2. So, when A wants to call
someone outside, it Dials to "C". "C" analyzes the "extension number"
and redirects it to the appropriate destination at "B", billing the
call:

A (exten 223) calls extension 978 at C <----> "C" knows extension 978 is

"B" extension 10978 and calls it  <-----> "B" accepts the call to 10978
from "C"

When connection between "C" and "B" is estabilished, "C" starts native
bridge mode, transfering call control. For "C", call ended and it bills
as it longs only few seconds.

Should I disable native bridge? How? I need "C" bills the call and
controls it.

Thanks in advance,

Daniel


--__--__--

Message: 9
Date: Fri, 12 Mar 2004 23:15:05 +1100
To: asterisk-users at lists.digium.com
From: Peter Brown <peterabrown at froggy.com.au>
Subject: Re: [Asterisk-Users] E1 cards in Australia
Reply-To: asterisk-users at lists.digium.com

Alex,

With Digium's agreement, I am certifying the TE410P for use in
Australia.

If you want please talk to me.

At 21:57 12/03/04 +1100, you wrote:
>Sorry for double post. Wrong subject :-)
>
>
>Hi All,
>
>Does anyone have Digium E1 cards in production in Australia? Are any of
them
>certified?
>Any feedback would be appreciated.
>
>Thaks
>Alex.
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

Peter Brown
CEO
IP Telephonics Ph 02 9153-5978



--__--__--

Message: 10
Date: Fri, 12 Mar 2004 05:26:13 -0600
From: Rich Adamson <radamson at routers.com>
Subject: Re: [Asterisk-Users] PCI front mount chassis?
To: asterisk-users at lists.digium.com
Reply-To: asterisk-users at lists.digium.com


> > I too am running 6 cards in my system, although not in a "high
traffic
> > capacity" load environment.
> >
> > So far my (limited) high-load simulations have shown no problems.
>
>
> So - is it apocryphal that the Digium cards (drivers) won't share
> interrupts?
>
> If there is a real issue with sharing interrupts then it seems to me
> to be a bug that needs fixing.  PCI bus supports shared interrupts,
> why doesn't the hardware/driver?

In most cases, sharing an interrupt is not a problem at all. There have
been a few cases where _some_ issue was resolved by moving cards around,
however the majority of those seem to be: a) abrupt system changes with
no effort to seriously identify the root-cause, b) newbie installations
where the condition of the underlying system infrastructure is totally
unknown, or, c) wild recommendations that might have had some basis a
long time ago but no longer apply.

Example: 'cat /proc/interrupts'
  9: 1854652239          XT-PIC  ehci-hcd, eth0, wcfxo, Intel ICH4
works just fine, and I can't imagine a more demanding irq arrangement
where the only nic shares with an x100p, etc.

Obviously there are performance limits and expecting multiple quad T1
cards or some other _specific_ high-volume configuration to share one
or two interrupts could create a problem. But, engineering a system for
those conditions is no more difficult then understanding the
requirements of whatever cards are being used and dealing with them
appropriately.




--__--__--

Message: 11
From: "David J Carter" <david.carter at codepipe.com>
To: "Asterisk User Group" <Asterisk-Users at lists.digium.com>
Date: Fri, 12 Mar 2004 12:42:33 -0000
Subject: [Asterisk-Users] Help on two subjects
Reply-To: asterisk-users at lists.digium.com


Hi All,


I have now got my '*' server up and running quite good.

As stated in earlier posts I am no Linux guru, so a bit of hand holding
required.

  First Subject.

I would now like to add h323 boxes to the '*' server, I have looked
through
the wiki and followed the instructions about what I need but I am a
little
thick as I can't seem to get to grips with it. Has anybody got a dummies
step by step guide to installing things needed for h323.

ala
1. turn on your server.
2. log onto your server.
3. make a cup of coffee because ya gonna need it.
4. ......
and so on.

   Second Subject.

I have never used or seen a channel bank, but I think it is what I
require
for a project I am looking at.

I have 12 Analogue (CO) lines that I would like to bring into the '*'
server.
I have 12 Analogue POTS that I would like to connect to the '*' server,
these are along with SIP phones (Grandstream), and IAX clients. The
later
two I have no problems with, see First Subject for the other failings.

If any one can help then please either answer on or off list.


Regards & thanks in advance.


Dave



--__--__--

Message: 12
Date: Fri, 12 Mar 2004 14:51:01 +0200
From: Michael Manousos <manousos at inaccessnetworks.com>
Organization: inAccess Networks
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] asterisk-oh323, new version 0.5.10
Reply-To: asterisk-users at lists.digium.com


T.38 FAX is in the short-term plans for asterisk-oh323.

Michael

T. Chan wrote:
> Dear Michael
>
> Do you foresee implementing these in the near future, one or the other
or
> both? Thanks
>
> Tc
>
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Michael
> Manousos
> Sent: Thursday, March 11, 2004 4:49 AM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] asterisk-oh323, new version 0.5.10
>
>
>
> Hi TC,
> T.38 FAX and native bridging are not supported by asterisk-oh323.
>
> Michael.
>
>
> T. Chan wrote:
>
>>Dear Michael,
>>
>>Does your H323 driver run T38 Fax? Also, does your H323 driver have
the
>>capability of just proxying signal, and NOT proxying signal and media,
>
> just
>
>>like the canrevite=yes in the sip scenario? Thanks
>>
>>TC
>>
>>-----Original Message-----
>>From: asterisk-users-admin at lists.digium.com
>>[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Michael
>>Manousos
>>Sent: Wednesday, March 10, 2004 7:11 AM
>>To: asterisk-users at lists.digium.com
>>Subject: [Asterisk-Users] asterisk-oh323, new version 0.5.10
>>
>>
>>
>>Hello all,
>>
>>asterisk-oh323 has been updated. The new version 0.5.10 fixes
>>the incorrect answering of H.323 channels (thanks to the people
>>of the list who helped to trace the problem). Also, I have added
>>support for Gnomemeeting text messages (just for fun).
>>Additionally, the new version contains stability improvements.
>>
>>This will be the last version using the OpenH323/Pwlib v1.12.2/1.5.2.
>>The next version will move on to the latest versions of these
>>libraries.
>>
>>Regards,
>>Michael.
>>
>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>---
>>Incoming mail is certified Virus Free.
>>Checked by AVG anti-virus system (http://www.grisoft.com).
>>Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004
>>
>>---
>>Outgoing mail is certified Virus Free.
>>Checked by AVG anti-virus system (http://www.grisoft.com).
>>Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004
>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> ./M
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ---
> Incoming mail is certified Virus Free.
> Checked by AVG anti-virus system (http://www.grisoft.com).
> Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004
>
> ---
> Outgoing mail is certified Virus Free.
> Checked by AVG anti-virus system (http://www.grisoft.com).
> Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
./M


--__--__--

Message: 13
Date: Fri, 12 Mar 2004 14:53:03 +0200
From: Michael Manousos <manousos at inaccessnetworks.com>
Organization: inAccess Networks
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] asterisk-oh323
Reply-To: asterisk-users at lists.digium.com


Hi,

Check the included README file for installation instructions.

Michael

Erick Weber V. wrote:
> Hi all:
>
> Does someone can direct me to an asterisk-oh323 how to or installation
> manual
>
> Thanks
>
> Erick
>
>


--__--__--

Message: 14
Date: Fri, 12 Mar 2004 13:12:12 +0000
From: stan <stan at saticed.me.uk>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] XML Phone book software.
Reply-To: asterisk-users at lists.digium.com

On Thu, Mar 11, 2004 at 04:06:41PM -0600, Brian R. Swan wrote:
> I'm looking into writing a some phone book XML/PHP software for my
Cisco
> phones.  Specifically, I'd like to be able to use a web interface (on
the
> computer) to maintain a contact list, and then dial from it on the
phone.
> Maybe using MySql on the back end or something (to be determined).
Before I
> start, and duplicate something else that exists, I wanted to see if
anyone
> has heard of software like that?  Searches of Sourceforge, Freshmeat,
and
> Google didn't turn up much or anything.
>

see the cmxml software section of
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx



--__--__--

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