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 ?

--=20
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 !

--=20
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=20
billing control. All connections are IAX-2. So, when A wants to call=20
someone outside, it Dials to "C". "C" analyzes the "extension number"=20
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=20
from "C"

When connection between "C" and "B" is estabilished, "C" starts native=20
bridge mode, transfering call control. For "C", call ended and it bills=20
as it longs only few seconds.

Should I disable native bridge? How? I need "C" bills the call and=20
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=20



--__--__--

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=20
> > capacity" load environment.
> >=20
> > So far my (limited) high-load simulations have shown no problems.
>=20
>=20
> So - is it apocryphal that the Digium cards (drivers) won't share
> interrupts?
>=20
> 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=20
cards or some other _specific_ high-volume configuration to share one=20
or two interrupts could create a problem. But, engineering a system for
those conditions is no more difficult then understanding the=20
requirements of whatever cards are being used and dealing with them=20
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
>=20
> Do you foresee implementing these in the near future, one or the other
or
> both? Thanks
>=20
> Tc
>=20
>=20
> -----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
>=20
>=20
>=20
> Hi TC,
> T.38 FAX and native bridging are not supported by asterisk-oh323.
>=20
> Michael.
>=20
>=20
> T. Chan wrote:
>=20
>>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,
>=20
> just
>=20
>>like the canrevite=3Dyes 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
>=20
>=20
> --
> ./M
>=20
> _______________________________________________
> 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
>=20
> ---
> 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
>=20
> ---
> 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
>=20
> _______________________________________________
> 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

--=20
./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:
>=20
> Does someone can direct me to an asterisk-oh323 how to or installation
> manual
>=20
> Thanks
>=20
> Erick
>=20
>=20


--__--__--

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=20
> phones.  Specifically, I'd like to be able to use a web interface (on
the=20
> computer) to maintain a contact list, and then dial from it on the
phone. =20
> Maybe using MySql on the back end or something (to be determined).
Before I=20
> start, and duplicate something else that exists, I wanted to see if
anyone=20
> has heard of software like that?  Searches of Sourceforge, Freshmeat,
and=20
> Google didn't turn up much or anything.
>

see the cmxml software section of
http://www.voip-info.org/tiki-index.php?page=3DAsterisk+phone+cisco+79xx



--__--__--

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Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users


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