asterisk 1.4.18.1 for AsteriskNow.
benoit
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
just arrived.
However this doesn't happens everytimes
Is it normal ?
regards,
benoit
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http
Atis Lezdins a écrit :
Calls are distributed in Priority+FIFO. Do you set ${QUEUE_PRIO}
before sending call to queue? Perhaps you're forgetting it in some
part of dialplan.
Thanks for the hint, could be usefull, however i'm not using it anywhere
right now
Philipp Kempgen a écrit :
Benoit schrieb:
I'm having a question with asterisk queue system, is it a fifo or a lifo
or random ?
Depends on the strategy.
http://www.voip-info.org/wiki-Asterisk+call+queues
Philipp Kempgen
? The strategy is for call distribution to member
Atis Lezdins a écrit :
On Thu, Dec 18, 2008 at 8:50 PM, Darrin Henshaw dhens...@ignition.bm wrote:
I believe you are correct Atis.
Philipp within your queue setup do you have any announcements? If so read
the posting on
Atis Lezdins a écrit :
You could enable core set verbose 3 and core set debug 1, and then
post corresponding log when you see this happens.
Ok so, what you are saying is that this shouldn't happens ? (in normal
conditions
the queue should be a fifo)
Anyway, i had the idea this was caused
nik600 a écrit :
Hi to all.
I'm using Asterisk 1.4 with Sjphone as softphone.
My problem is that when a SIP user is busy, he still receive calls
from asterisk.
I've tried to setup the call-limit preference to 1, but with this kind
of configuration the user can't transfer calls, as the
queues, is there one or should i manually maintain a
list ?
regards,
benoit
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman
of their shift,
they must do so. Putting in place an automated soluition will make
them sloppier, so be sure this is something you can control.
Just my two eurocents,
l.
2008/12/22 Benoit maver...@maverick.eu.org
mailto:maver...@maverick.eu.org
Hi,
To force user to behave
Hi,
I'm migrating some macro from extension.conf format to AEL and they are
some things
i don't understand, and some i don't evan know how to get them working.
Example:
Doing this queueMemberList=${QUEUE_MEMBER_LIST(queue)}; won't
make any warning
message, however the resulting
Benoit a écrit :
Hi,
..
Also, there is this construct which isn't working:
if( ${DB_EXISTS(family/key)} ) {
} else {
}
However i can't find a workaround in this case ... any idea ?
Well, never mind ths one, some space got stuck in the macro
Hi,
After seeing in pbx/ael/ael-test/ael-test5/extensions.ael some
interesting use case of RealTime
to store extension data (forwardto, dnd, ...) i started to play with it.
To my surprise the two applications RealTime() / RealtimeUpdate() have
been deprecated in favor
of the REALTIME()
Benoit a écrit :
Hi,
After seeing in pbx/ael/ael-test/ael-test5/extensions.ael some
interesting use case of RealTime
to store extension data (forwardto, dnd, ...) i started to play with it.
To my surprise the two applications RealTime() / RealtimeUpdate() have
been deprecated in favor
Olivier Fauchon a écrit :
Hi.
When I call my RNIS numbers (with a mobile phone for example), I can
see 2 incoming calls on the IPBX, which should not happend.
I'm not sure if it's a problem with the telco France Telecom and their
ISDN setup, or if it's a problem
with the MISDN driver on
Ariel Dorfman a écrit :
Hi all
This is my first post.
As the subject says, I need to implement on my call center the Agent
functionality, son the agents could logon and logoff to the queue
How can I do this configuration? Or where can I read some info about it
Regards
Ariel
To
Brent Davidson a écrit :
Watkins, Bradley wrote:
...
Well, before I file a bug I have another question... In AEL,
what is the correct syntax? Do all variable references still need to be
wrapped in ${} or not? If they do, then the documentation on
voip-info.org needs to be
Daniel Varella a écrit :
Hi everybody,
Happy New Year !
I'm trying to detect if a call was answered by a machine (linke
voicemail systems) or a human.
I would like to use AMD (Answering Machine Detect) command, but
with my configuration it was not possible get there.
Follow
I don't think your problem is somehow related to the debian release ...
However since mISDN 1.1.8 was released to support kernel 2.6.24 25 the
18/06/2008
and the kernel 2.6.26 was released the 13/07/2008 an incompatibility
between both
could very well be possible.
Well and it's quite simple,
Hi,
I'm a little surprised, up until 1.4.22 my agents where using an IAX
channel to ZoIPer Softphone,
however since after the upgrade to .22 we experienced a problem with
hangup failure between zoiper
and asterisk (look like bug http://bugs.digium.com/view.php?id=13184) i
made them switch to
Isn't anyone using this kind of thing http://www.red-fone.com/ for this
kind of massive HA deployment ?
there is a case study on their web site using 4 asterisk boxes and 8
red-fone T1 to Ethernet bridges to
handle 900 concurrent calls.
(I would say that using all this nice hardware on Linksys
Mark Michelson a écrit :
Benoit wrote:
Hi,
I'm a little surprised, up until 1.4.22 my agents where using an IAX
channel to ZoIPer Softphone,
however since after the upgrade to .22 we experienced a problem with
hangup failure between zoiper
and asterisk (look like bug http
Olivier a écrit :
2009/1/9 Philipp Kempgen philipp.kemp...@amooma.de
mailto:philipp.kemp...@amooma.de
Olivier schrieb:
Before diving into this, I would very pleased to know if someone
could yes
or no, successfully compile mISDN 1.1.8 on Lenny (latest RC1 or
beta2
Olivier a écrit :
Hi,
While issuing make, I've got:
...
CC [M] /usr/src/zaptel-1.4.11~dfsg/kernel/zaptel-base.o
/usr/src/zaptel-1.4.11~dfsg/kernel/zaptel-base.c: In function
âzt_registerâ:
/usr/src/zaptel-1.4.11~dfsg/kernel/zaptel-base.c:5230: error: implicit
declaration of function
Olivier Fauchon a écrit :
Hi Benoit,
Thx for your previous reply.
I tired your configuration , and I still have the problem . When 1
make a call to my SDA(DDI) number, I can see 2 incoming calls in the
isdn stack.
What do you think about that ?
P[ 2] channel with stid:0 for one
I've given a shot at res_snmp, which does include some interesting
datas, however i see the following
problem:
The astChanTypeTable if fully dynamic (one can have SIP as row 2
while another at 12, i don't even know if
the sequence is persistent within one's setup). To create
nice
Something like has been discussed a few day ago, i think you need to
remove the host=
string and add username/password.
Right now asterisk may allow you request from the autorized IP ranges,
but the authentification
of the request fail due to the invalid host. you need to switch to
Well, at worst you can make use of the exec and pass_persist feature of
snmpd
that way you can build you own script that will run locally to the
asterisk box
and query using snmp
Grygoriy Dobrovolskyy a écrit :
I wonder if the same is possible with centreon ?
Someone is using centreon here ?
Personnaly, i had recently encountered a global machine check exception
with
two cards (TE220p and B410) and many kernel panic with mISDN (mostly if
i tried to unload it).
Dahdi still hasn't failed me (directly)
Thomas Kenyon a écrit :
Yesterday, a low-duty production server that I maintain
Hi,
Our potentiel next phone provider ask me a question i can't answer for sure,
maybe someone here knows ?
He says that is equipement only support VN4 protocol or more, or ETSI,
however i can't find matching terms in the digium documentation or
the chan_dahdi/dahdi/system.conf files...
Any
Laurent a écrit :
Those terms would be ISDN-related. VN4 is Version Number 4, and
ETSI is the European standards-adopting organization for telecoms.
So you might want to check for E1 support (ISDN in Europe,
basically) if you want to connect a PRI-capable equipment - I
assume that's what you
Benoit a écrit :
Laurent a écrit :
Those terms would be ISDN-related. VN4 is Version Number 4, and
ETSI is the European standards-adopting organization for telecoms.
So you might want to check for E1 support (ISDN in Europe,
basically) if you want to connect a PRI-capable equipment - I
Laurent a écrit :
...
I read somewhere (can't remember where, sorry) that double
framing is equivalent to basic framing. Presumably it is also
equivalent to no crc. If that's correct, then I would say you
can use the same setup as you have with FT, it looks good to me.
Ok, thanks
As
Laurent a écrit :
Le 19.01.2009 08:50, Benoit a écrit :
Laurent a écrit :
Well, the telcos techs said a straight cable should do the trick, but
since i didn't get any isdn link up
with the straight, i built a crossover like what you described, with no
luck either
Laurent a écrit :
Le 19.01.2009 08:50, Benoit a écrit :
Laurent a écrit :
Well, the telcos techs said a straight cable should do the trick, but
since i didn't get any isdn link up
with the straight, i built a crossover like what you described, with no
luck either
Benoit a écrit :
Laurent a écrit :
Did you check (like with a multimeter or something similar) the
connectivity of your cable ? the first E1 crossover cable I made
had a problem (entirely my own fault) and I thought it didn't
work. The way I checked was by connecting the two ports
I now this page has something like that:
http://www.softwink.com/papers/Installation_Securing_VoIP_With_Linux/
but on a PRI and not a BRI, but maybe it can be extrapolated
Lee Wilson a écrit :
Its been a few days, I was wondering if anyone else has any ideas on how to
get this to work?
If
I second that, while read an berkeley db file outside of it's main
application
can work fine, writing in it would certainly lead to huge trouble (data
loss, corrupted file, ...)
A berkeley db file is .. a file, not a database server
David fire a écrit :
external DB? like mysql?
2009/1/24
Olivier a écrit :
From my point of view, the most important feature is TE-PTMP as this
the one used here (in France) when connecting a new Asterisk-based
IPBX to ISDN (I really don't know why TE-PTP is not used for that).
Well this may depend of some parameter. We have a dual BRI line from
Olivier a écrit :
Anyway, according your own experience, how frequent is this PTP case ?
I can't say, i don't have any other experience than our own lines
That's the point : as much as possible, we're trying to avoid any
re-configuration of legacy equipment.
So, how would you order TE/NT,
f...@hotbox.ru a écrit :
Hi everyone!
I've set up asterisk ip-pbx to implement IVR menu and encountered such a
problem: when users dial the destinaion phone number and end it up with
# asterisk still waits until timeout in WaitExten() is reached.
Well i don't see anything in the doc
extra line
could do not harm.
If not, this would be nice to be fixed :)
(Asterisk 1.4.23.1)
regards,
benoit
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit
Here is my current setup:
E1 = [Asterisk with TE220p] = IAX Trunk (routed network) =
[Asterisk with TDM800p] = Fax/Copy Machine
This seems to work fine, a few failed Fax or very slow sending/process
sometimes but no complaining
users, so this must be ok :)
My previous try was:
E1 =
Remco Barendse a écrit :
On Sun, 8 Mar 2009, benoit wrote:
Here is my current setup:
E1 = [Asterisk with TE220p] = IAX Trunk (routed network) =
[Asterisk with TDM800p] = Fax/Copy Machine
The TE220P and the TDM800P are in different Asterisk boxes? Any particular
reason
Doug Lytle a écrit :
benoit wrote:
Remco Barendse a écrit :
On Sun, 8 Mar 2009, benoit wrote:
Here is my current setup:
E1 = [Asterisk with TE220p] = IAX Trunk (routed network) =
[Asterisk with TDM800p] = Fax/Copy Machine
})});(adding
of double quote)
Is it normal ?
benoit
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com
Le 10/01/2010 07:53, Tilghman Lesher a écrit :
On Saturday 09 January 2010 15:22:29 Benoit wrote:
I'm playing around with asterisk 1.6.2.0 and the first try was to
replace my now non-functionning
'app-realtime' macro which emulated RealTime with REALTIME_HASH()
There is very few
Le 12/01/2010 16:35, Tilghman Lesher a écrit :
On Tuesday 12 January 2010 04:44:36 Benoit wrote:
I just experienced another problem however i have two rnis cards, one
b410p and one te220,
while the later works prefectly i can't really make the first one work,
using DAHDI or mISDN
under
Le 13/01/2010 09:57, Benoit a écrit :
Le 12/01/2010 16:35, Tilghman Lesher a écrit :
On Tuesday 12 January 2010 04:44:36 Benoit wrote:
I just experienced another problem however i have two rnis cards, one
b410p and one te220,
while the later works prefectly i can't really make
On 22/01/2010 19:10, Benoit wrote:
Le 13/01/2010 09:57, Benoit a écrit :
Le 12/01/2010 16:35, Tilghman Lesher a écrit :
On Tuesday 12 January 2010 04:44:36 Benoit wrote:
I just experienced another problem however i have two rnis cards, one
b410p and one te220,
while
On 20/02/2010 01:35, Daniel Bareiro wrote:
alderamin*CLI
-- Executing [...@from-internal:1] Dial(SIP/danib-089f8820,
SIP/300|30|tTrm) in new stack
[Feb 19 19:22:50] WARNING[19254]: app_dial.c:1237 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone
Hi,
I'm using an half T1 line on a asterisk (obviously :)) 1.6.2.4 system,
up to recently everything
was fine but we are starting to experience the call limitation of the
line (15).
So as to warn user of the problem i attached a vocal notification to the
CONGESTION status after a Dial(),
but it
Le 22/02/2010 09:28, Conor McTernan a écrit :
I've just encountered an odd problem with our Digium TE410P card and
was wondering if anyone has experienced something similar before.
There is one similar request on this list from a few weeks back iirc
We quickly removed the card and checked the
On 12/06/2010 15:09, sean darcy wrote:
I decided to include the following in each sip.conf stanza that has an
outgoing context:
deny=0.0.0.0/0.0.0.0
permit=10.10.10.0/24
If all your phones are on a defined network like that, you really should
use iptables to allow
inbound SIP from the
Hi,
For some reason (outbound call tracking) I've got a few different
outbound call process (using a macro for queuemetrics logging, or direct
call)
i wanted to factorise the routing process so i came up with something
like the following. All in one it's working like expected, however
every
Le 23/07/2010 16:44, Zeeshan Zakaria a écrit :
Hi,
I try to avoid any warnings, as they can turn into errors later.
well, that's exactly the point of this inquiry :)
I remember having problems with GoSub long time ago, don't remember
what it was, but I moved to macros after that.
For
Hi,
I'm quite pleased with the asterisk/res_snmp integration (at least a
right one :) not some hackish scripted thingy)
but i felt it's missing quite a few datas.
What i would need is:
* Per channels, number of inbound call received since asterisk
startup (like a network interface)
*
Hi,
I'm experiencing an issue with asterisk 1.6.2.10 12rc1,
i'm not sure if it's to be expected or not, so here it is:
When transferring call (blind-transfer) using asterisk feature key,
things are working OK, however when using ZoIPer's transfer key
(which is implemented with a Refer-To SIP
On 27/10/2010 12:59, Krzysztof Urbaniak wrote:
Hi!
We've experienced asterisk has gone without any message, it wasn't any
segfault, anything in asterisk messages log that says about shutting
down.
How do you launch asterisk ? did you try without 'safe_asterisk' or
anything like it,
just
Le 28/10/2010 08:41, Krzysztof Urbaniak a écrit :
2010/10/27 Benoitmaver...@maverick.eu.org:
On 27/10/2010 12:59, Krzysztof Urbaniak wrote:
Hi!
We've experienced asterisk has gone without any message, it wasn't any
segfault, anything in asterisk messages log that says about shutting
down.
On 07/11/2010 19:29, Cary Fitch wrote:
I don't want to start the How many calls can Asterisk handle? discussion
or How many angels can stand on the point of a pin? discussion either.
But can anyone contribute some practical knowledge of systems that take in
channel bank T1s or DS3s from far
to be dialed by an unknown party this way)
Regards,
benoit
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http
setup a livevoip DID and indeed the DTMF does not work.
The same asterisk context works via broadvoice and via
direct dialing in to the asterisk server via SIP.
Just no DTMF with calls via livevoip.
I'm running Asterisk CVS-v1-0-03/06/05-23:15:12
--
Mike Benoit [EMAIL PROTECTED
it
does) while this tends to take a while with VoIPJet.
I've also seen 'circuit fast busy' message - what is the difference
between the two?
Thanks,
Jean-Michel.
--
Mike Benoit [EMAIL PROTECTED]
signature.asc
Description: This is a digitally signed message part
Asterisk-Users@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
--
Mike Benoit [EMAIL PROTECTED]
signature.asc
Description: This is a digitally signed message part
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Benoit
Sent: Thursday, April 07, 2005 7:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Liveviop problem
Speaking of LiveVOIP and there west coast server
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Mike Benoit [EMAIL PROTECTED]
signature.asc
Description: This is a digitally signed message part
___
Asterisk-Users mailing list
Asterisk-Users
mailing list
Asterisk-Users@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
--
Mike Benoit [EMAIL PROTECTED]
signature.asc
Description: This is a digitally signed message part
/listinfo/asterisk-users
--
Mike Benoit [EMAIL PROTECTED]
signature.asc
Description: This is a digitally signed message part
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
/mailman/listinfo/asterisk-users
--
Mike Benoit [EMAIL PROTECTED]
signature.asc
Description: This is a digitally signed message part
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi All,
I realize there has been much talk about Asterisk web interfaces in the
past, and there are even a few created such as vmail.cgi and Astweb.
However, coming from the perspective of trying to convince upper
management of a large multinational company to convert its entire phone
, or are you using a codec other than u-law
or A-law? Sometimes the slow control messages will get through the wrong
modem, but the fast modem for the images never will.
Regards,
Steve
Mike Benoit wrote:
I'm trying to send a fax to my asterisk box, however shortly after
connecting the fax
code 1
On Thu, 2004-06-17 at 13:33 -0700, Mike Benoit wrote:
Good question Steve. My setup is basically:
Fax Machine - PSTN - X100P - Asterisk - RxFax
I'm not even sure, does Asterisk do encoding if its not sending the call
to a SIP device, or over IAX?
In the mean time I configured
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
musiconhold=default
faxdetect=both
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
--
Mike Benoit [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED
-users
--
Mike Benoit [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Mike Benoit [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL
of getting the right
technician on the phone?
Thanks.
--
Mike Benoit [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http
]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Mike Benoit [EMAIL PROTECTED
, 2004-06-30 at 22:42 -0500, Daniel Jimenez wrote:
Mike Benoit wrote:
So whats the next step?
How much money are you willing to put in the project?
Are you talking POTS lines or a PRI?
If this is a serious project and you'd really like to clear it up I'd
look at a Cisco device (maybe
calls are made from SPA-2000's to PSTN numbers
through Asterisk, asterisk is just amplifying the SPA-2000's own echo
somehow.
Any test results are welcome, I would be very interested if other people
are unable to replicate my results.
--
Mike Benoit [EMAIL PROTECTED
://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Mike Benoit [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman
After trying everything under the sun to get rid of echo on my X100P,
I'm curious if anyone managed to solve the echo issues by switching to a
SPA-3000?
As well, if you have multiple SPA-3000's, can you create dial-out groups
similar to the Dial(ZAP/g1) functionality?
Thanks.
--
Mike Benoit
this point the finger at
Asterisk code having issues?
On Wed, 2004-06-30 at 16:36 -0700, Mike Benoit wrote:
Over the last couple weeks I've tried everything I could get my hands on
in an attempt to get rid of my echo problems. Using a CVS checkout of
just yesterday, I've tried every echo
at 02:06 -0400, Anton wrote:
No it points to Cell phone companies having better hardware echo
cancellation on their lines, also cell phones themselves have a hardware
echo can built in.
- Original Message -
From: Mike Benoit [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July
visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Mike Benoit [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http
Dameon and Wolfgang,
Have either of you experienced echo when making a call from the FXS
port to the FXO port on the SPA-3000?
Thanks
--
Mike Benoit [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com
in
to something like parkedcalls. But for now, the above works relatively
well. The biggest drawback is there is no way to get back to call on
hold until it times out.
Enjoy.
--
Mike Benoit [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED
/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Mike Benoit [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo
: 1.562500
Is this a poor mainboard issue, or is it actually not possible to do IDE
software RAID on a machine running Asterisk with X100P cards?
Is anyone currently doing it?
Thanks.
--
Mike Benoit [EMAIL PROTECTED]
___
Asterisk-Users mailing list
://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Mike Benoit [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http
the X100P card, or the TDM400 for
that matter I'm interested if it has the same issue, and run the same
test. See if you notice any cutting out.
--
Mike Benoit [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com
/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Mike Benoit [EMAIL PROTECTED]
signature.asc
Description: This is a digitally signed message part
___
Asterisk-Users mailing list
[EMAIL
thing here. A bad SPA?
--
Tim Schacher
[EMAIL PROTECTED]
218-844-5985
On Tue, 2004-11-30 at 12:31, Mike Benoit wrote:
This sounds similar to the issue I have been dealing with for the past
several months.
Are the calls just being dropped, or is the SPA rebooting itself? One
/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Mike Benoit [EMAIL PROTECTED]
signature.asc
Description: This is a digitally signed message part
___
Asterisk-Users
, Brian Roy wrote:
On Wed, 01 Dec 2004 16:01:16 -0800, Mike Benoit [EMAIL PROTECTED] wrote:
Don't run LISa on the same network as any SPA-2000 or SPA-3000. (maybe
even any Sipura device?)
I have a problem with mine locking up, but not while talking. When it
sits idle for a period of time I come
installed:
gcc-3.3.2-9mdk
gcc-cpp-3.3.2-9mdk
Any ideas how to fix this?
Thanks.
--
Mike Benoit [EMAIL PROTECTED]
signature.asc
Description: This is a digitally signed message part
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http
--
Mike Benoit [EMAIL PROTECTED]
signature.asc
Description: This is a digitally signed message part
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Mike Benoit [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE
:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Mike Benoit [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http
/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Mike Benoit [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo
/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Mike Benoit [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo
1 - 100 of 289 matches
Mail list logo
