Hello Thorsten,
i had a trace with core set debug 10 and core set verbose 10 but i
didn't find anything usefull.
The log is very full so it could be that i missed some important information.
This is a the less verbose output of the problem:
-- SIP/22-01b3 answered SIP/64-01b2
--
Currently I have no idea. But you wrote, that it does not happen all the
time. Please provide us a log extract from that case where is going
wrong. Perhaps you can do a diff for the good and the bad case
yourself before?!
Am 02.10.2012 09:02, schrieb Gianluca Baù:
Hello Thorsten,
i had a
If this is the wrong place to post this I'm sure someone will let me know. :-)
I'm looking for a reliable, inexpensive call termination service
(SIP). The one I am presently with does not seem to know what IPs they
send inbound calls from, and it is maddening to keep up with the FW
changes
On Tue, Oct 2, 2012 at 10:44 AM, Chris Nighswonger
cnighswon...@foundations.edu wrote:
If this is the wrong place to post this I'm sure someone will let me know.
:-)
I'm looking for a reliable, inexpensive call termination service
(SIP). The one I am presently with does not seem to know
On Mon, Oct 01, 2012 at 06:44:22PM -0400, Pat Collins wrote:
Can anyone tell me if it is possible to invert the signaling bits
on a T1 channel?
I need to emulate PLAR signaling in asterisk. EM seems to be an
exact match if reversed.
I need idle bits and seized
Perhaps you could
Thank you for the reply!
So far, I've managed to get the on off hook to work properly!
My next problem is the incoming ring. I've changed the /include/dahdi/user.h
file:
#define DAHDI_ABIT (1 3)
#define DAHDI_BBIT (1 2)
#define DAHDI_CBIT (1 1)
#define
Niccol
How about to change tone list on indications.conf file?
Please comment call waiting line ; according to country zone or default
settings.
Good luck
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
--
_
-- Bandwidth and
On Mon, Oct 1, 2012 at 10:03 AM, Niccolò Belli darkba...@linuxsystems.itwrote:
Hi,
The call waiting tone is very annoying (you hear nothing while it plays
the beep). I need callwaiting because of the queues (the phone has to ring
as soon as you hangup) but I want to remove the beep on my
On 10/1/2012 5:19 PM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Michelson
Sent: Monday, October 01, 2012 4:15 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
Hi,
Il 02/10/2012 21:04, Mc GRATH Ricardo ha scritto:
How about to change tone list on indications.conf file?
As I already said indications.conf doesn't work for dahdi channels,
unfortunately the callwaiting tone is hardcoded in asterisk itself (not
even in dahdi/libtonezone!). I solved
So a few people just reported that they couldn't make any calls. I
logged into asterisk and at first everything on the console looked
normal, then I got swamped with messages about too many open files.
This is from my asterisk/messages log file:
[Oct 2 16:46:00] WARNING[19429] rtp.c: Unable
Thank you, I already considered such an approach but the customer wanted
to receive the new call *immediately* after the hangup (basically
because it was possible with the old pbx).
This is how I solved: http://www.spinics.net/lists/asterisk/msg153399.html
Such a way I hear the annoying beep
I was looking at open files. lsof | wc -l tells me around 2000 or so.
The total number goes up and down, but hovers around 2000 and doesn't
seem to show any upward trend. I haven't rebooted the system since I
killed and restarted asterisk, so the first guess would be that asterisk
is what
Adam Moffett писал 03.10.2012 00:09:
manager.c: Accept
returned -1: Too many open files
It's not a problem of old Asterisk.
You should check your system limits with the ulimit -a
You can
increase different limits, say ulimit -n 4096 will increase limit for
open files to 4096
I suggest
On 2/10/12 6:51 pm, Carlos Alvarez wrote:
Your traffic level, number of concurrent calls, etc would help us know what
sort of carrier you should be talking to.
Equally important, your geographic location, and the geographic location
to which most of your calls are made will be useful in
I'm looking at what would be involved in converting from MeetMe to
ConfBridge and there seems to be a lot of missing administrative things,
but I hope I'm just missing it. We all know about the missing realtime
linkage. That's a major nuisance, but can be worked around.
More serious is that the
NB == Niccolò Belli darkba...@linuxsystems.it writes:
NB If someone knows how to COMPLETELY REMOVE the fucking beep please
NB let me know: there are already tons of phones ringing everywhere so
NB there is no need for an annoying beep.
Edit chan_dahdi.c.
The my_callwait() and/or
Asterisk 1.8 on Ubuntu
We store the configuration files in CVS. We have a development, QA and
production environments. 90% of the config files are the same across all
3 environments, but there are some differences in sip.conf and
extensions.conf (environment specific voip providers and/or
On Tue, Oct 2, 2012 at 5:30 PM, Chris Bagnall
aster...@lists.minotaur.cc wrote:
On 2/10/12 6:51 pm, Carlos Alvarez wrote:
Your traffic level, number of concurrent calls, etc would help us know
what
sort of carrier you should be talking to.
Equally important, your geographic location, and
On 12-10-02 06:39 PM, Mitch Claborn wrote:
Asterisk 1.8 on Ubuntu
We store the configuration files in CVS. We have a development, QA and
production environments. 90% of the config files are the same across all
3 environments, but there are some differences in sip.conf and
extensions.conf
On Tue, 2 Oct 2012, Mitch Claborn wrote:
I'd like to be able to use the same config files in CVS and have the
differences resolved at run time, based on host name of the asterisk
server.
Another idea would be to write a simple perl or other program to
pre-process the files and put some
On Tue, Oct 2, 2012 at 8:04 PM, Steve Edwards asterisk@sedwards.com wrote:
On Tue, 2 Oct 2012, Mitch Claborn wrote:
I'd like to be able to use the same config files in CVS and have the
differences resolved at run time, based on host name of the asterisk server.
Another idea would be to
At 02:19 PM 10/1/2012, you wrote:
So respond here and let me know what you think. I got a couple of replies on
the -dev list and they said that this would be good to put out on the -users
list too.
Mark Michelson
In true Republican fashion, I'm going to vote for case-insensitivity.
Given
On 10/1/2012 4:15 PM, Mark Michelson wrote:
Hi!
I've been confronted with an interesting issue to resolve. The
issue is located here:
https://issues.asterisk.org/jira/browse/ASTERISK-20163
The issue involves case-sensitivity of channel and global variables
in the dialplan. Current
- Original Message -
From: Vladimir Mikhelson v...@mikhelson.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, October 2, 2012 9:02:18 PM
Subject: Re: [asterisk-users] Case-sensitivity of Dialplan variables.
On
On Mon, Oct 1, 2012 at 4:15 PM, Mark Michelson mmichel...@digium.comwrote:
Hi!
I've been confronted with an interesting issue to resolve. The
issue is located here:
snip
So respond here and let me know what you think. I got a couple of replies
on the -dev list and they said that this
Hola,
Richard Kenner wrote:
I'm looking at what would be involved in converting from MeetMe to
ConfBridge and there seems to be a lot of missing administrative things,
but I hope I'm just missing it. We all know about the missing realtime
linkage. That's a major nuisance, but can be worked
- Original Message -
From: Ira i...@extrasensory.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, October 2, 2012 8:11:32 PM
Subject: Re: [asterisk-users] Case-sensitivity of Dialplan variables.
Given that many of the
Shaun,
To make more sense of the code, I changed
#define DAHDI_XBIT (3 2) to
#define DAHDI_XBIT (0)
Sadly, incoming calls do not work. Not sure exactly how to START or RING
when the RX AB bits are 00
Any ideas?
Thanks again for your help!
-Original Message-
On 10/2/2012 9:12 PM, Warren Selby wrote:
On Mon, Oct 1, 2012 at 4:15 PM, Mark Michelson mmichel...@digium.com
mailto:mmichel...@digium.com wrote:
Hi!
I've been confronted with an interesting issue to resolve. The
issue is located here:
snip
So respond here and let
2012/10/1 Mark Michelson mmichel...@digium.com
Hi!
I've been confronted with an interesting issue to resolve. The
issue is located here:
https://issues.asterisk.org/**jira/browse/ASTERISK-20163https://issues.asterisk.org/jira/browse/ASTERISK-20163
The issue involves case-sensitivity of
On Tue, Oct 02, 2012 at 11:22:31PM -0400, Pat Collins wrote:
Shaun,
To make more sense of the code, I changed
#define DAHDI_XBIT(3 2) to
#define DAHDI_XBIT(0)
Sadly, incoming calls do not work. Not sure exactly how to START or RING
when the RX AB bits are 00
32 matches
Mail list logo
