Hi,
I did replace an old asterisk box by a new shiny one (2 BRI ports used
on a quad port card - BeroNet PCI Express)
I noticed a message in the logs that puzzles me:
May 11 19:10:02 kernel: dahdi: Master changed to B4/0/1
May 11 19:10:08 kernel: wcb4xxp :05:04.0: PCI INT A disabled
May 11
On 08/06/2010 19:19, Steve Edwards wrote:
> The ONLY way (how's that for humble) to do this in a reliable and robust
> method is to use a real database. Personally, I like MySQL and I prefer to
> do database work in an AGI in a compiled language like C.
>
> Maintaining the accumulated duration in a
On 08/06/2010 15:21, Danny Nicholas wrote:
> My .02 - I would set up a context for dialing with this provider that counts
> minutes and stops the dial with a message once you get to 1321 minutes (22
> hours).
>
> Exten => _8X.,1,noop(call using Cheap sip)
> Exten => _8X.,2,macro(call_out,${EXTEN:
Hi,
I'm currently using a cheap SIP provider for outbound calls.
I do have 6 channels to them.
In their terms of service there is the following limit:
The total duration of calls during one single day should not exceed 24
hours or we do have the right to terminate the contract...blah blah
Wha
On 31/01/2010 04:35, Tzafrir Cohen wrote:
> Yes, please see https://issues.asterisk.org/view.php?id=16493
>
> Basically the driver needs minimal fixing. Probably just to add the PCI
> ID to the list.
>
Hi,
Thanks
Gonna have a look at it.
--
_
Hi,
I'm currently trying to get a BN4S0e (which is basically a BN4S0 with a
PCIe connector) working with dahdi.
The module is loading properly but the card is not detected by the module.
Is support on dahdi planned for this card ?
In the meantime i'm gonna use mISDN with this card.
Thanks
La
Laurent CARON wrote:
> I'm experiencing a quite strange behavior while trying to receive faxes
> through Asterisk (either directly through app_rxfax or with spandsp +
> hylafax).
Hi,
I should have mentionned (known?) that the telco is using G729 compression.
Obviously FAXes
Hi,
I'm experiencing a quite strange behavior while trying to receive faxes
through Asterisk (either directly through app_rxfax or with spandsp +
hylafax).
Config:
HFC quad BRI card (3 T0 connected to the card)
Asterisk 1.4.21
asterisk-app-fax 0.0.20070624-2
hylafax 2:4.4.4-10.1
libpri 1.4.2
li
Hi,
Since we're moving from a legacy PABX that has been serving one
of our customers for more than 15 years, we'd like this process to
require no "human habits" change among the users.
Software: Asterisk 1.4.22
Hardware: Polycom phones (mainly 430/601)
Here are the "problems":
We did configure c
Hi,
I'm sorry i didn't check the recipient while replying.
Sorry about the noise...
Laurent
Le 11 nov. 08 à 11:44, Laurent Caron <[EMAIL PROTECTED]> a écrit :
> Bonjour Louis-David,
>
> Asterisk envoie-t-il le signal au boitier pour
Bonjour Louis-David,
Asterisk envoie-t-il le signal au boitier pour le failover ?
Laurent
Le 11 nov. 08 à 08:49, Louis-David Mitterrand <[EMAIL PROTECTED]
g> a écrit :
> Hi,
>
> I tried "upgrading" from debian's 1.4.21.2 package to vanilla 1.4.22
> but
> then my TE410P alarms stay RED and
Hi,
We did move an office from a remote building to another floor in our
building, allowing us to directly hook the switch in that building to
our core switch (GigE).
In the past, the phones were on the same subnet as the * server.
All the phones worked flawlessy for about two years.
Since t
Hi,
I'm experiencing a strange behavior on one of my asterisk servers.
When I make a sip connection (sip.conf) between the 2 boxes using the
primary interface it works fine.
When however use an aliased interface (ethX:Y) it fails.
from sip show peers:
ast-rem-vpn 192.168.3.253
Hi,
One of my users is in trouble with his polycom phone hooked to an
asterisk server.
The phone works fine for a few days, and then disappears from the
registered sip peers in asterisk.
The user is able to place outbound phone calls, but can't receive
incoming calls until the network plug is un
Rob Schall wrote:
> Auto-congesting is a "inconclusive" error message from what I've found.
> In this case, it probably means, "I haven't heard from that phone or the
> response ping failed", etc.
>
> So in this case, I'd say you're right, the phone probably is giving up,
> and isn't trying to re-
Rob Schall wrote:
> In the logs, does that phone try to re-register itself, or does it just
> give up?
The phone is giving up.
Jun 7 14:29:36 NOTICE[22015] chan_sip.c: Auto-congesting
SIP/XXYYZZAA24-08553940
Laurent
___
--Bandwidth and Colocation prov
Rob Schall wrote:
> In the logs, does that phone try to re-register itself, or does it just
> give up?
>
> If its not trying to re-register, you might want to look at the
> "Expires", "Register" and Retry settings in the phone.
>
Here is the config snippet:
Thanks
Laurent
__
On Thu, May 03, 2007 at 12:47:46AM +0200, Laurent Caron wrote:
> On Sun, Apr 29, 2007 at 09:06:53PM +0200, Clayton Milos wrote:
> > Since a PRI is a physical connection as well as a logical one, if you can
> > get the server to shut down when it has a problem you could put a 4-pol
On Sun, Apr 29, 2007 at 09:06:53PM +0200, Clayton Milos wrote:
> Since a PRI is a physical connection as well as a logical one, if you can
> get the server to shut down when it has a problem you could put a 4-pole
> relay to change the PRI over to the other box.
>
I think the ISDNGuard is more o
Tim Panton wrote:
> Here's what we do when consulting in this area:
>
> First decide what the maximum acceptable downtime is, and
> what the costs to the business of that downtime would be.
>
> Use that as the starting point for the HA design.
>
> Discuss with the telco what they can do.
They c
Hi,
I'm wondering what the best option to obtain a high availability
asterisk server is.
I currently use a TE410P (4 x E1) card.
I'm thinking of 2 different solutions:
- 2 servers configured with Heartbeat + DRBD (drbd mainly for
voicemail) and the E1 span plugged to the 2 servers (with a
Hi,
I've got a serious problems.
I have an * box set up at a custommer office.
* seems to work well until this message appears when i try to call from
the outside to any number managed by *
Jul 7 17:12:08 WARNING[8792]: chan_zap.c:9256 pri_dchannel: Ring
requested on channel 0/1 already in
chawki hammoud wrote:
hi:
i would greatly appreciate it if somebody can refer me
to asterisk consultants.
Hi,
Here you are
http://www.voip-info.org/wiki-Asterisk+consultants+Europe
Laurent
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