hen i call, the script is executed and the call goes in queue, i can hear
the MOH, the file /tmp/pippo is created but it is empty.
Any idea or suggestion?
PS:
if i use the application monitor or MixMonitor the call is recorded
correctly.
I'm using Asterisk 1.6.2.9-2+squeeze12
Thanks
--
/
On Fri, 1 Jul 2016, nik600 wrote:
>
> i've tried rfc2833,inband and info having the same behaviour in all
>> situation.
>>
>> 2016-06-30 23:53 GMT+02:00 nik600 <nik...@gmail.com>:
>> sorry for top-posting, the two topics started with 2 different
>> reaso
ito,,1)
exten =>01,n,SayDigits(${digito})
Any idea?
2016-07-01 0:13 GMT+02:00 nik600 <nik...@gmail.com>:
> i've tried rfc2833,inband and info having the same behaviour in all
> situation.
>
> 2016-06-30 23:53 GMT+02:00 nik600 <nik...@gmail.com>:
>
>> sor
i've tried rfc2833,inband and info having the same behaviour in all
situation.
2016-06-30 23:53 GMT+02:00 nik600 <nik...@gmail.com>:
> sorry for top-posting, the two topics started with 2 different reason
> subject, but then we finished on the same problem.
>
> btw,
e only difference i see is the "1st File Descriptor" pointing to -1
2016-06-30 23:29 GMT+02:00 Steve Edwards <asterisk@sedwards.com>:
> Please don't top post.
>
> On Thu, 30 Jun 2016, nik600 wrote:
>
> this is the point, and the strange thing:DTMF is s
Kiniston <johnkinis...@gmail.com>:
> Looking at your logs it looks like you may need to modify your sip.conf,
> Check with your provider as to what kind of DTMF they support and configure
> sip.conf to use that type of signalling.
>
>
>
> On Thu, Jun 30, 2016 at 1:18 P
i'm using Asterisk 1.6.2.9-2+squeeze12
2016-06-30 22:14 GMT+02:00 Richard Mudgett <rmudg...@digium.com>:
>
>
> On Thu, Jun 30, 2016 at 3:00 PM, nik600 <nik...@gmail.com> wrote:
>
>> Dear all
>>
>> i'm creating an outgoing call to number xxx with this c
t; n,Playback(pls-wait-connect-call)
> same => n,MacroExit();Return
>
> exten => REJECT,1,NoOP()
> same => n,Playback(beep)
> same => n,Set(MACRO_RESULT=BUSY);Reject the call
> same => n,
bx2-04ad",
"digito,,1") in new stack
[Jun 30 21:56:56] WARNING[5617]: channel.c:2558 ast_waitfordigit_full:
Unexpected control subclass '-1'
-- User entered nothing.
Any idea?
if i call from number xxx to an extension that goes to testDTMF@cRETEUNICA
it works p
oh, yes!
Many thanks
2016-06-30 15:28 GMT+02:00 Guido Falsi <m...@madpilot.net>:
> On 06/30/16 15:08, nik600 wrote:
> > Dear all
> >
> > i'm using an "old" Asterisk 1.6.2.9-2+squeeze12, and want to know if is
> > possible to configure a scenario l
(?)
Step 1,2,3 works properly but i'm not able to link the two channels, even
using redirect,goto or pickupChan.
Any idea or help will be appreciated!
Thanks
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93m 5596 S1 3.1 0:00.68 asterisk
15956 root 20 0 662m 93m 5596 S1 3.1 0:00.68 asterisk
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ignores the | delimiter, if i try with the
comma it works.
Reading the the upgrade file it seems that the pbx_realtime should
affect also the extension.conf settings... where am i wrong?
Thanks to all in advance
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i've also tied this tests:
- changed hardware
- upgrade to 1.4.31
- kernel recompiled with 1000 Hz option
- changed SO (Slackware 13)
- run the system on hardware (no ESXi)
But i've not resolved the problem.
Do you have any idea?
On Thu, May 6, 2010 at 11:54 AM, nik600 nik...@gmail.com wrote
i get may debug messages like this:
DEBUG[30684] channel.c: Internal timing is disabled
(option_internal_timing=0 chan-timingfd=-1)
Is because dahdi is not installed?
Can this be a possible cause of this behaviour?
On Tue, May 4, 2010 at 9:54 PM, nik600 nik...@gmail.com wrote:
Dear all
setinterfacevar=yes
eventwhencalled=yes
eventmemberstatus=yes
ringinuse=no
member = SIP/PL1039
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I've resolved...it was a limitation of the provider for calls without a CallerID
On Sun, Apr 18, 2010 at 7:43 PM, nik600 nik...@gmail.com wrote:
Dear all
i'm trying to originate an outgoing call with the command originate,
from Asterisk's CLI i'm typing:
CLI originate IAX2/my-iax-provider
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Many thanks Jonathan!
On Wed, Mar 31, 2010 at 10:29 AM, cov...@ccs.covici.com wrote:
What is the significance of /dev/fd/3 where does it come from?
I'ts the file descriptor 3 for the EAGI process, wich contains the audio.
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but the recording file is continuosly growing and ffmpeg
ends the conversion before of the call completion.
If you can give me a practical example i'll appreciate it a lot.
Bye
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users information and thansfer call.
Do you know if there is something similar somewhere ?
Maybe Asterisk has already some magic sauce to do that ? ;-)
Thanks to all
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this kind of approach
- if someone else has done something similar and wants to share his experience
- how much is affordable the events generation excpecially in system
with a high load
Thanks to all for any contribute.
Hi
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I've tested and confirm that the AGI script can do that.
i had to enable setinterfacevar=yes in the queue conf and then can
read the MEMBERINTERFACE channel variable.
Just because it can be useful for someone else.
On Fri, Oct 23, 2009 at 9:44 PM, nik600 nik...@gmail.com wrote:
Hi to all
an appropriate AGI script can i play an audio file (or
create it with some tts) to the call?
After the AGI script the call is linked with the operator even if
there is an Answer into the AGI?
Thanks to all
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-complete-2.2.0.2
libpri-1.4.10.1
Any idea?
On Tue, Oct 13, 2009 at 11:51 PM, nik600 nik...@gmail.com wrote:
for disabling the hardware DTMF you intend to recompile zaptel with
vpmdtmfsupport=0?
Thanks
On Fri, Oct 9, 2009 at 6:54 PM, C F shma...@gmail.com wrote:
are you using chan_local
for disabling the hardware DTMF you intend to recompile zaptel with
vpmdtmfsupport=0?
Thanks
On Fri, Oct 9, 2009 at 6:54 PM, C F shma...@gmail.com wrote:
are you using chan_local?
try disabling the hardware DTMF.
Sent using my wired Blueberry.
On 10/9/09, nik600 nik...@gmail.com wrote
Hi to all, is it possible to setup a live audio streaming in Asterisk
using for source monitor, mixmonitor or chanspy?
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On Fri, Oct 9, 2009 at 6:54 PM, C F shma...@gmail.com wrote:
are you using chan_local?
try disabling the hardware DTMF.
Sent using my wired Blueberry.
On 10/9/09, nik600 nik...@gmail.com wrote:
Dear all
i have a TE205P connected to an Asterisk 1.2.18.
Yes i know, the version is old but since
as i get the wrong dtmf tone
directly from Asterisk.
It's not a phone problem as the same phone may retry and then it works.
Is it possible to relate it with the load of the server?
Can you suggest me something?
Thansk
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any interest in it?
I'm evauating to add this feature but before to do that i'd like to
know if there is some other approach that can avoid some developement.
Regards
On Wed, Sep 30, 2009 at 12:48 PM, nik600 nik...@gmail.com wrote:
Dear all
is it possible to handle a queue using a programmed
)
And then manually match information between unique ID and queue_log to
consider info on queue A,B,C,D, as a single queue.
Or is there some magic sauce to specify an IVR script that is
executed when a call is in a queue?
Thanks
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?
My aim is to make a REFER to b...@test and free completely Asterisk.
Thanks to all in advance, bye.
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...
Bye
On Fri, Jun 12, 2009 at 2:45 PM, Giorgio
Incantalupogincantal...@fgasoftware.com wrote:
Hi nik600,
I had some trouble transferring calls with that version of Asterisk even
if I used the normal transfer via features.conf. Upgrading to 1.4.24
helped a bit (even if not completely). My advice
/ configuration to use a complete and stable
implementation of the REFER functionality?
Thanks to all in advance
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, 2009 at 3:57 PM, Florian Hackenberger
f.hackenber...@chello.at wrote:
On Sunday 08 March 2009 17:11:33 nik600 wrote:
Hi to all isn't there any plan to add the Skills Based Routing
strategy in queues.conf?
I think that it will be enough to add an int skill to the struct
member and then order
difference.
Is there something else that i can do?
Thanks to all
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on each
dialplan step or is better to parse the logfile and extract the
information needed?
I'm using Asterisk 1.4.23.1
TIA
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On Thu, Mar 12, 2009 at 8:13 PM, Matt Riddell li...@venturevoip.com wrote:
On 13/03/2009 8:02 a.m., nik600 wrote:
Hi to all.
What can i do if a customer needs to log in the CDR all the dialpan
actions related to a call?
I mean, not only the lastapp e the lastdata but all the dialpan actions
application call...
well, this will be surely the best.
I'll read the documentation and let you know, thanks.
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, can you tell me where you have to place the code to
log when an app is called?
Thanks
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On Tue, Mar 10, 2009 at 4:01 AM, James Sneeringer jsnee...@gmail.com wrote:
On Mon, Mar 9, 2009 at 5:44 PM, nik600 nik...@gmail.com wrote:
Thanks, i've tested and it works (1.4.23.1).
Just 2 questions:
1) this approach seems to be an hack and not the implementation of a
feature is it really
that?
maybe i should open a new post but i think that this kind of approach
isn't much better than the callback functionality, what do you think
about that?
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Thanks for your time.
Bye
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Hi., do you think that sbr policy in queue strategy will be useful?
Bye
-- Forwarded message --
From: nik600 nik...@gmail.com
Date: Sat, 7 Mar 2009 15:21:14 +0100
Subject: add a new queue strategy: SBR
To: Asterisk Developers Mailing List asterisk-...@lists.digium.com
Hi to all
but priority are se to the call, not to the agent!
or am i wrong?
On Sun, Mar 8, 2009 at 5:32 PM, David fire ddf...@gmail.com wrote:
the queue already have prioritys.
David
2009/3/8 nik600 nik...@gmail.com
Hi., do you think that sbr policy in queue strategy will be useful?
Bye
it.
The rtp traffic is redirect correctly but the SIP INVITE contains the
ip of the lan and not of the nat.
I'll try with SipAddHeader and then let you know...
thanks
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On Sat, Feb 7, 2009 at 8:31 AM, nik600 nik...@gmail.com wrote:
hi
is it possible to set up in the dialplan (on in sip.conf, or something
else) the hostname of the outgoing uri call?
This is my scenario:
- CCM integrated with Asterisk via h323
- SIP user registerd to Asterisk
- Asterisk
) the call is forwarded to
x...@10.10.10.2 that is the wrong address.
I've tried to force SIP_HEADER(CONTACT) in the dialplan with a Set but
it seems that i can't due to security reason.
Is it possible to avoid this problem?
Thanks
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problem.
The question is: can one server with those settings manage up to 200
simultaneous call?
The server will receive SIP calls and forward them through a CISCO router.
Thanks to all
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otherwise.
It means that rfc2833 was offered, but doesn't work!
Well, info and inband works.
Bye
On Thu, Jan 22, 2009 at 11:18 AM, nik600 nik...@gmail.com wrote:
Is there the possibility to increase the debug of an AJAM command?
If DTMF works on channel, and my command is queued successfully
Is there the possibility to increase the debug of an AJAM command?
If DTMF works on channel, and my command is queued successfully, what
can be the problem?
Thanks
On Thu, Jan 15, 2009 at 4:34 PM, nik600 nik...@gmail.com wrote:
Hi to all
i'm using PlayDTMF with AJAM, after the authentication
' //response
/ajax-response
But i can't heard nothing on the channel, i've tried to send the tone
both on channel and link, but with no results.
If i use normal dtmf from keyboards they works properly.
What can i check?
Thanks
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need DAHDI services, you must correctly configure DAHDI.
Where am i wrong?
Thanks
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PS:
asterisk is compiled with dahdi support
On Mon, Jan 12, 2009 at 1:39 PM, nik600 nik...@gmail.com wrote:
Hi to all.
I'm trying to use meetme on asterisk 1.4.22.1.
On a debian i've compiled (as i need h323 support)
openh323_v1_18_0
pwlib_v1_10_0
dahdi-linux-2.1.0.3
dahdi-tools
hi to all.
Do you know if there is an asterisk 1.4 package with h323 support for debian?
I've found this http://packages.debian.org/etch/asterisk-h323 but has
asterisk 1.2.13.
Thanks to all.
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sorry if i ask it again, but where can i find the patch for enable
busy-level/limit in 1.4 ?
thanks
On Tue, Nov 18, 2008 at 12:09 PM, nik600 nik...@gmail.com wrote:
Thanks, is it possibile to retrieve a patch from Asterisk trunk? how?
On Tue, Nov 18, 2008 at 11:54 AM, Steve Howes st
Ok, i've resolved, the problem was related to the sip type settings.
It must be peer instead of fried.
Bye
On Fri, Jan 2, 2009 at 5:41 PM, nik600 nik...@gmail.com wrote:
Thanks for your reply.
Now, i use devstate too, but it doesn't work (or, maybe i suppose that
it should work differently
the user wants
waiting calls or not and decide accordingly.
__Yehavi:
2008/12/20 nik600 nik...@gmail.com
On Sat, Dec 20, 2008 at 2:25 PM, Benoit maver...@maverick.eu.org wrote:
Have you tried to set the call-limit to 10 or 2 for example, i know it's
what's needed
method to limit the call available for a user to 1
but granting him the possibility to transfer a call?
I know that there is the busy-level settings, but i'ts available only in 1.6.
Thanks to all in advance.
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(correctly...up to 2 calls), even when he is busy.
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Hi
using AMI, is it possile to stream a file on a specific channel?
Thanks to all in advance.
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do it from a SIP protocol perspective. I'm not sure to what extent Asterisk
supports this capability.
--
Raj Jain
ok, thanks for your reply!
I'll search about Asterisk SIP referer implementation.
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Maybe my question is not clear or is too stupid? (sorry)
Maybe this is already done in SIP trunking?
Or (worste case) is impossible to do that?
Thanks
On Fri, Nov 21, 2008 at 8:53 AM, nik600 [EMAIL PROTECTED] wrote:
Hi to all.
i-ve got a question:
what happen when a call between 2 trunks
- Caller3
or
b) Caller 1 - Trunk A/C - Caller3
So:
is it possible to avoid the scenario a) ?
Thanks to all
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before registration
limitonpeers = yes
call-limit = 2
busy-level = 1
The directive busy-level is ignored
I've also tried busy-limit but without any result...
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Thanks, is it possibile to retrieve a patch from Asterisk trunk? how?
On Tue, Nov 18, 2008 at 11:54 AM, Steve Howes [EMAIL PROTECTED] wrote:
On 18 Nov 2008, at 10:30, nik600 wrote:
the busy-level / busy-limit setting in sip.conf is available for
Asterisk 1.4.22 ?
http://www.voip-info.org
Hi to all.
Is possible with the Asterisk 1.4 cli view the current calls and their codec?
Thanks to all
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.
(218)727-4332 x105
- nik600 [EMAIL PROTECTED] wrote:
Hi to all.
Is possible with the Asterisk 1.4 cli view the current calls and
their codec?
Thanks to all
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- nik600 [EMAIL PROTECTED] wrote:
And if i have an h323 configuration?
Thanks
On Tue, Nov 11, 2008 at 4:17 PM, Tim Nelson [EMAIL PROTECTED]
wrote:
[EMAIL PROTECTED] ~]# asterisk -rx 'sip show channels'
assuming you want SIP... substitute sip for iax2 if you prefer...
Tim
Hi to all
except of some commercial hardware / software gateways, is there any
opensource or free project to setup a Skype Account on Asterisk?
Thanks to all
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Hi what version of openh323 and pwlib are suggested for asterisk
1.4.21.1.? Thanks to all
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/libpthread.so.0
#12 0xb731905e in clone () from /lib/libc.so.6
Can someone help me please?
Thanks in advance to all
On 7/17/08, Patrick [EMAIL PROTECTED] wrote:
On Thu, 2008-07-17 at 19:34 +0200, nik600 wrote:
Hi what version of openh323 and pwlib are suggested for asterisk
1.4.21.1.? Thanks to all
Hi to all
is it possibile (via AMI or dialplan) to disable the DTMF tone on a
particular channel?
Thanks in advance
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i've seen that there is the PlayDTMF command.
Bye
On Tue, Jun 24, 2008 at 8:37 AM, nik600 [EMAIL PROTECTED] wrote:
any idea?
On Sat, Jun 14, 2008 at 9:50 AM, nik600 [EMAIL PROTECTED] wrote:
Hi to all
can i play a sound or a dtmf tone on a specific channel using AMI?
Thanks to all
any idea?
On Sat, Jun 14, 2008 at 9:50 AM, nik600 [EMAIL PROTECTED] wrote:
Hi to all
can i play a sound or a dtmf tone on a specific channel using AMI?
Thanks to all
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https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https
Hi to all.
How can i retrieve the status of a user using the subscription?
For example, if i use:
exten = 200,hint,SIP/200
exten = 200,1,Dial(SIP/200)
After that, how can i retrieve the status of the SIP/200 user using AMI ?
Thanks to all in advance
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https
Hi to all
can i play a sound or a dtmf tone on a specific channel using AMI?
Thanks to all
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will hear (and Asterisk can detects, via
AGI or dialplan) 200,300,400 DTMF tones.
You can find more information here.
http://www.kumbe.it/pagine/dettaglio/34/206.html
Bye
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https
on a single machine, i intend an
enterprise SX infrastructure with multiple nodes and auto failover
policy.
If Asterisk doens't suffer a virtualization, a service virtualized on
a solid infrastructure is more scalable and hardware independent
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to determine what is beta and what is stable?
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On Thu, May 22, 2008 at 12:31 PM, Gordon Henderson
[EMAIL PROTECTED] wrote:
On Thu, 22 May 2008, nik600 wrote:
Hi to all
i'm managing a call center with 20 operators using Asterisk.
I'm still using Asterisk 1.2.x as i love his stability.
Now, i'm planning to migrate to 1.4.x, but i don't
Hi to all
is it possible to retrieve the sip tag (server side) of a sip call
from the dialplan?
Thanks.
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https://sourceforge.net/projects/reportmaker
?
Thanks
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nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
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asterisk
On Dec 24, 2007 8:07 PM, Darrick Hartman [EMAIL PROTECTED] wrote:
Tzafrir Cohen wrote:
On Mon, Dec 24, 2007 at 05:11:44PM +0100, nik600 wrote:
maybe i've guess the problem!
on the same server, i've got a B800P.
I've tried to manually remove all isdn module and zaptel modules
On 12/23/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Dec 23, 2007 at 07:05:37PM +0100, nik600 wrote:
Hi
i've got an openvox a800p01 with 1 FXO and 4 FSX
i've done the following:
- downloaded zaptel-1.4.7.1
- downloaded the file opvxa1200.c
- copied in zaptel-1.4.7.1
maybe i've guess the problem!
on the same server, i've got a B800P.
I've tried to manually remove all isdn module and zaptel modules.
After that, i've done
modprobe zaptel
modprobe opvxa1200
and now the card has been correctly registered!
On Dec 24, 2007 2:32 PM, nik600 [EMAIL PROTECTED
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nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
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asterisk-users mailing list
interface
Subsystem: Unknown device 9100:0001
Flags: bus master, medium devsel, latency 32, IRQ 10
I/O ports at a800 [size=256]
Memory at f800 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2
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nik600
/mISDN-1_1_7/drivers/isdn/hardware/mISDN/stack.c:235
st(0100) addr(41000100) layer -1 out of range
* DMESG
Can you help me to guess the problem?
Thanks
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nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https
problems with
misdn drivers.
Thanks
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nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
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nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
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asterisk-users mailing list
for blind transfer!
Many thanks!
On Nov 20, 2007 2:24 PM, Atis Lezdins [EMAIL PROTECTED] wrote:
nik600 wrote:
Hi
i've read this post
http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html
I just want to know if there are some upgrades... on 1.4 or 1.2.
I'd like
instructions for the creation of
queue_stats table
- added the files view.sql
bye
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nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
for
each iax account?
Thanks
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nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser
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