Any particular reason you're using H323 instead of SIP ?
>
>
> _
>
> Darren Sessions
> [EMAIL PROTECTED]
> http://www.darrensessions.com
> _
>
>
>
>
>
> On Sep 16, 2008, at 12:04 PM, Guilherme Loch
I have a Cisco 3845 with a ISDN PRI port connected to my legacy PBX, this
router is running IOS 12.4(5) T5. I'm trying to integrate Asterisk with this
router through H.323, I tried ooh323 (comes with asterisk-addons) and it
works partially, I can make calls from Cisco to Asterisk, but the other way
a change-log, but I highly doubt a
> serious modification to the gsm code took place between sub-versions.
> Hope this helps,
>
> - Darren
>
>
> _
>
> [EMAIL PROTECTED]
> http://www.darrensessions.com
> http://www.linkedin.com/in/dsessions
>
I have a server with Asterisk 1.4.21.1 and some prompts recorded in GSM
format. I have these same prompts in another server with Asterisk 1.4.18, on
this server the prompts sound pretty nice, but on the first one they sound
pretty choppy. Was there any changes on the transcoding code between this 2
Hi,
I have a Xorcom Astribank connected to my Asterisk server. In one of the
Astribanks FXO port I have a Celular Interface Module. My problem is the
Astribank is receiving a early answer from the module, which doesn't happen
with a ATA connected to the same module. This is causing some trouble wit
Runnig the xpp_fxloader before the Zaptel and Asterisk scripts solves the
problem.
thank you Tzafir.
On Thu, May 8, 2008 at 11:07 AM, Tzafrir Cohen <[EMAIL PROTECTED]>
wrote:
> On Thu, May 08, 2008 at 08:33:08AM -0300, Guilherme Loch Waltrick Góes
> wrote:
> > Tzafir,
>
loadzone = us
defaultzone = us
Any hints ? What else can I do ?
Best Regards,
On Wed, May 7, 2008 at 4:23 PM, Tzafrir Cohen <[EMAIL PROTECTED]>
wrote:
> On Wed, May 07, 2008 at 01:03:53PM -0300, Guilherme Loch Waltrick Góes
> wrote:
> > I'm using Zaptel 1.4.10 compiled fro
Bus 001 Device 001: ID :
[EMAIL PROTECTED]:~# zaptel_hardware
[EMAIL PROTECTED]:~#
On Wed, May 7, 2008 at 11:18 AM, Tzafrir Cohen <[EMAIL PROTECTED]>
wrote:
> On Wed, May 07, 2008 at 09:20:59AM -0300, Guilherme Loch Waltrick Góes
> wrote:
> > I'm trying to use a Xor
I'm trying to use a Xorcom Astribank wth Ubuntu 8.04, but got no success. I
can see the channel bank with lsusb, but when I tried to use
zaptel_hardware, or when I try the /etc/init.d/script, they don't see my
Channel Bank. I compiled the latest Zaptel 1.4.10, with Astribank's
dependecies, fxload a
This happens because the TOS and DSCP are the same field. TOS is the first
implementation of QoS on the IP header, DSCP is it's evolution and uses the
same field on the IP header, you can use only one of the two at the same
time.
Best Regards,
On Wed, May 7, 2008 at 8:59 AM, Vikas <[EMAIL PROTECTE
What is the default username/password. In the Maestro forum's it only says
it's hardcoded, but doesn't say the actual username/password.
Best Regards,
On Tue, Apr 15, 2008 at 4:43 PM, Lee Jenkins <[EMAIL PROTECTED]> wrote:
> Darryl Dunkin wrote:
> > FOP works for us, no need for X:
> > http://www
I have a server with 8 Xorcom Astribank, they're connected to the server via
a USB hub, my problem is: when a reboot the server the ID of the Channel
Bank changes, cousing a big mess on my server. Anybody know how can I solve
this ?
Best regards,
--
Guilherme Loch Góes
Visite nossa loja virtual:
y
> protection is based on the MAC of the interfaces in the system.
>
> Guilherme Loch Waltrick Góes wrote:
> > I'm trying to use the Digium suplied G.729 Codec, I have ran the
> register
> > utility, and got my licenses written to /var/lib/asterisk/licenses, but
>
I'm trying to use the Digium suplied G.729 Codec, I have ran the register
utility, and got my licenses written to /var/lib/asterisk/licenses, but when
a start Asterisk I got the following errors:
[Mar 24 12:33:08] NOTICE[4068] codec_g729a.c: G.729 transcoding module
version 34, Copyright (C) 1999-
Asterisk-java, http://asterisk-java.org is a very good one, it has a pretty
good documentation.
On Thu, Mar 6, 2008 at 1:01 PM, equis software <[EMAIL PROTECTED]>
wrote:
> Hi, I need to interact with my Asterisk and need a good Java class
> library.
> What do you think is the best?
>
> Thanks
>
>
I have an Asterisk server with voicemail(), in the sip.conf I have:
[general]
allowguest=yes
language=pt_BR
I have the sound files for pt_BR in /var/lib/asterisk/sounds/pt_BR, and the
others dirs (dgits, phonetic and so on). The problem I have is: when a guest
tries to place a call and is directed
e to control
> the entire call from beginning to end, or can I just call the Asterisk-Java
> class at certain points with certain parameters?
>
>
>
> Thanks!
>
>
> Evan
>
>
>
>
>
>
>
>
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL P
Have a look at asterisk-java.org. I has everything you need.
On Jan 29, 2008 4:35 PM, Evan Ruff <[EMAIL PROTECTED]> wrote:
> Hey Guys,
>
>
>
> I've been doing some research into the AGI-Java connector and was
> wondering if somebody could help me with my architecture.
>
>
>
> What I'd like to d
You could also use a call file ( google it ;) ).
On Jan 18, 2008 6:37 PM, LWATCDR <[EMAIL PROTECTED]> wrote:
> I would like to add a function to an existing application that will
> make an outgoing call.
> I found this example using the Manager API for originating a call to
> an extension.
>
>
>
On zapata.conf use the parameter callerid.
On Jan 17, 2008 3:33 AM, sandeep <[EMAIL PROTECTED]> wrote:
> hi all,
> how to set the caller id facility for
> the TDM400p card.
>
> Please help me
>
> thanks,
> sandeep.s
>
> --
Guilherme Loch Góes
Visite nossa loja virtual: http://www.shopvoip.com.b
Yes, I use it and got no problems.
On Jan 16, 2008 11:47 AM, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> Hi All;
>
> Did anyone tried to use dns name or ddns name with
> host (host=abc.www.com) and it worked fine?
>
> Regards
> Bilal
>
>
>
>
>
Change the priority of the second dial() to 4.
Regards,
On Jan 16, 2008 11:42 AM, Abdul <[EMAIL PROTECTED]> wrote:
> Good Day All,
>
> Is it possible to put backup route in asterisk dial plan? fro the example
> if the first carrier disconnect the call with Congestion or Circuit busy
> then asteri
Please expliain more, show us your extensions.conf.
On Jan 16, 2008 9:51 AM, Rahul Yadav <[EMAIL PROTECTED]> wrote:
> Hi all
> This is rahul i am using asterisk 1.4.17 with degium TE120p card on PRIE.
> I have configured everything card but there is a problem coming
> asterisk is dialing _98XXX
What's the difference between the TE121 and TE122. I read the description on
Digium's site and it isn't clear to me.
Best regards,
--
Guilherme Loch Góes
Visite nossa loja virtual: http://www.shopvoip.com.br
Notícias e Fórum sobre VoIP com software livre:
http://www.asteriskexperts.com.br
_
Have a look at the new Digium list: Asterisk-HA, I think this thread makes
more sense there.
Best Regards,
On Jan 15, 2008 4:42 PM, Jeremy Mann <[EMAIL PROTECTED]> wrote:
> Has anyone ever written asterisk logic to Heartbeat remote phone lines?
> Something that would dial out and see if a busy t
Lookup the automon feature on features.conf .
Best regards,
On Jan 15, 2008 4:55 PM, Anciso, Roy <[EMAIL PROTECTED]> wrote:
> Just wondering if this is possible:
>
> Make a call from a registered sip extension (Doesn't matter if it's
> internal or external) during the call press a key sequence l
Does anyone got Astribank working with AsteriskNOW beta 6 ? The bug in
mantis seems to be closed, but I cannot find "fxload" or "lsusb" to do some
debugging.
--
Guilherme Loch Góes
MSN:[EMAIL PROTECTED]
(48) 99115299
___
--Bandwidth and Colocation Prov
A little off topic, but SipX has built in redudancy. if it is so
important to you, you should have a look.
On 9/25/07, Atis Lezdins <[EMAIL PROTECTED]> wrote:
> [snip]
> > http://lists.digium.com/pipermail/asterisk-users/2007-August/195339.html
> > >> could provide you with some answers.
> > >
> >
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