On Mon, 2011-01-24 at 18:46 -0500, Richard Kenner wrote:
I'm confused about a few things relating to realtime, SIP and config in
general.
As I understand it, with the exception of extensions.conf, I can either
have a config file completely in text or completely in a database. Is
that
Try changing 'hostname=127.0.0.1' to 'hostname=localhost' in the
cdr_mysql.conf. I seem to remember a problem I had when '127.0.0.1' and
'localhost' didn't marry up never did find out why.
If that doesn't work - try GRANT SELECT , INSERT ,UPDATE ON
`Asterisk`.`cdr` TO 'asteriskcdr'@'localhost';
Hi Leif,
Submitted as requested - https://issues.asterisk.org/view.php?id=18672
Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif
Madsen
Sent: 21 January 2011 15:58
To:
Il 24/01/2011 16:43, Jonas Kellens ha scritto:
Hello list,
I keep on getting the error :
ERROR[1707] cdr_addon_mysql.c: Cannot connect to database server
127.0.0.1: (1045) Access denied for user 'asteriskcdr'@'localhost'
(using password: YES)
I have a 'cdr' table in my MySQL-DB. On this
On 25 Jan 2011, at 09:36, Andrew Thomas wrote:
Try changing 'hostname=127.0.0.1' to 'hostname=localhost' in the
cdr_mysql.conf. I seem to remember a problem I had when '127.0.0.1' and
'localhost' didn't marry up never did find out why.
I believe localhost means it can use a socket, where as
On Monday 24 Jan 2011, Jonas Kellens wrote:
I keep on getting the error :
ERROR[1707] cdr_addon_mysql.c: Cannot connect to database server
127.0.0.1: (1045) Access denied for user 'asteriskcdr'@'localhost'
(using password: YES)
[stuff deleted]
Try hostname=localhost aot hostname=127.0.0.1,
Thanks Dave. Sounds like a man who's not had his hand soaking in ivory
liquid and been through the toils and tortures of various upgrades over the
years. Very insightful though. Goof thing this discussion ensued as I am
learning a lot about what to be wary of not least of all, the truth about
also the Answer is redundant
in logger.conf console = warning,error,notice,debug,fax
2011/1/24 David Backeberg dbackeb...@gmail.com:
On Mon, Jan 24, 2011 at 2:53 PM, Bryant Zimmerman brya...@zktech.com wrote:
I am testing out inbound faxing using res_fax and res_fax_spandsp.so
My system
In my quest to resolve the ongoing issues I have with outgoing calls...
In the SIP invite, the SDP describes the media and RTP ports. If I run
sockstat I believe I can see the ports available, but if I run tcpdump I
see no packets pass or even get blocked at the firewall. How is it
initiated
On Tue, Jan 25, 2011 at 6:59 AM, Andrew Latham lath...@gmail.com wrote:
Thanks Dave. Sounds like a man who's not had his hand soaking in ivory
liquid and been through the toils and tortures of various upgrades over
the
years. Very insightful though. Goof thing this discussion ensued as I
On 01/24/2011 2:54PM Bryant Zimmerman wrote
I am testing out inbound faxing using res_fax and res_fax_spandsp.so My
system answers the call but then sets there on the ReseiveFax line then
comes back with an error that it exceeded the maximum retries. How
Yes, it is possible. Multiple peers can be registered by a single IP.
I have Snom 300 which supports 4 identities.
On Fri, Jan 21, 2011 at 2:28 PM, Warren Selby wcse...@selbytech.com wrote:
On Fri, Jan 21, 2011 at 12:53 PM, Olivier oza_4...@yahoo.fr wrote:
Hi,
Is it possible for a SIP to
On Tue, Jan 25, 2011 at 9:34 AM, Bryant Zimmerman brya...@zktech.com wrote:
On 01/24/2011 2:54PM Bryant Zimmerman wrote
The attached file was too large so I am putting in a link to the file. It is
a virus free text file.
You failed to mention earlier that this is T.38.
Turn off T.38 and see
Indeed, but the calling channel is Dahdi and it is connected to a POTS line.
2011/1/24 C F shma...@gmail.com
I guess that was the CallerID transmitted by the calling channel.
On Mon, Jan 24, 2011 at 7:31 PM, Jose Flores Galicia floj...@gmail.com
wrote:
Hi List.
Have any of you guys
Hi,
I'm having a problem trying to register sip trunks.
I using asterisk 1.4.39.1, freepbx 2.5.2 in centos 5.5 and I'm trying to
configure several sip accounts from my provider. The accounts are
individually configured as sip trunks.
With only one account everything is ok, it registers and I
On Tue, Jan 25, 2011 at 1:45 PM, Bryant Zimmerman brya...@zktech.com wrote:
Do you know how to force off T.38 in res_fax?
it's in sip.conf
take a look for
t38pt_udptl=yes
change it to no
reload sip
on your console
that should force it to either fail entirely or do audio passthrough.
--
Hi all,
i am running asterisk by using command asterisk -r, asterisk -vc
when i want to come out of asterisk it not getting exit or quit
from the shell
i have tried soo many options like...
stop now
stop gracefully
exit
stop
quit
its not working sti can any one
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
viswavardhanreddy karna
Sent: Tuesday, January 25, 2011 1:38 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] regarding quit, exit and stop now in asterisk
Hi
thanks danny
On Tue, Jan 25, 2011 at 8:43 PM, Danny Nicholas da...@debsinc.com wrote:
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *viswavardhanreddy
karna
*Sent:* Tuesday, January 25, 2011
OK, I generally use Hylafax+IAXModem for our faxing, but I have been
fiddling with FFA and SpanDSP for a while.
Is there a good way to determine what version of SpanDSP I have
installed and whether the app_fax.so module is the same version?
Many thanks,
Tom
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Rymes
Sent: Tuesday, January 25, 2011 2:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Help determining SpanDSP
From: David Backeberg dbackeb...@gmail.com
Sent: Tuesday, January 25, 2011 1:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ReceiveFAX issue.
On Tue, Jan 25, 2011 at 1:45
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