On 8/21/2010 6:09 PM, Zeeshan Zakaria wrote:
Then may be these big multi-billion dollar corporations should use one
of them, with whom we all deal regarding various services, and who put
us through these voice recognition time-wasting activity in a hope
that the poor caller will eventually
All,
I am contemplating moving static SIP users to SIP realtime, and I am
wondering if there is a nice simple tool to be able to do this with? I
am not concerned with something that would do all the work for me, just
something easier to use for a decent set of changes, than pure sql or
Steve Edwards wrote:
On Mon, 7 Jun 2010, bruce bruce wrote:
CentOS 5.4 and asterisk does stay running after it's loaded by asterisk
-g. But the chkconfig --add asterisk doesn't work :(
What does chkconfig --list asterisk show?
The add command looks in the asterisk script for a line
On 6/7/2010 5:20 PM, bruce bruce wrote:
Thanks for the input Seann and Steve. That is insightful. I did run
chkconfig --list asterisk and following is the output:
*[r...@tel ~]# chkconfig --list asterisk*
*asterisk0:off 1:off 2:on3:on4:on5:on6:off*
In file
The op_server.pl is part of the Flash Operators Panel, which isn't
really important to the operation of the PBX, it is just a nice pretty
interface showing what lines and what groups are active. What O/S are
you using? Are there any errors in the asterisk logs? Does asterisk stay
running after
the performance of an actual F5, or anything
like that.
Thanks in advance,
Seann Clark
smime.p7s
Description: S/MIME Cryptographic Signature
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New to Asterisk
clear idea of how to parse properly XML data in
either python or perl, but I have templates of what did work (until
formats changed, StormSiren being a python module I used for sms). Also
if I ever get anything to work, and anyone is interested I can share my
code.
Regards,
Seann Clark
All,
I just noticed this in my logs, and am rather lost as to what module
it pertains to. I would assume pseudo-realtime priority for the process,
but I am looking for a little confirmation from the group:
[Apr 28 12:28:36] WARNING[20773] asterisk.c: The canary is no more. He
has
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Seann Clark
Sent: Wednesday, April 28, 2010 2:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Strange Error -- ASterisk 1.6
All,
I just noticed this in my logs, and am
All,
I have been fighting with my dialplan for hours now, and google
searches talk lots but offer nothing in terms of explication for this. I
have my SIP peer set up and working with Broadvoice:
[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Seann Clark
Sent: Tuesday, April 27, 2010 7:15 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Broadvoice inbound fails on Asterisk 1.6.1
All
On 4/27/2010 8:07 PM, Richard Kenner wrote:
[sip.broadvoice.com]
...
[broadvoice]
exten = 551234,1,Set(CDR(accountcode)=44)
and Asterisk is still giving me this error in the logs (while playing a
number does not exist sound clip):
[Apr 27 18:11:19] NOTICE[12179]
On 4/26/2010 7:33 AM, Vieri wrote:
--- On Sun, 4/25/10, Gordon Hendersongordon+aster...@drogon.net wrote:
Hi,
I've noticed that one of my new servers (new mobo) if
drifting slowly
backwards in time (in aprox. 24 hours, system time
drifts back 5
minutes).
this is
happening. I can include any additional details if requested, as I don't
know exactly what would be helpful to others with this. The SPA itself
hasn't been changed in seven months, and is stable with Callweaver.
Thanks in Advance,
Seann Clark
smime.p7s
Description: S/MIME Cryptographic
this moment, but tomorrow I
should have time for this).
Thanks in advance,
Seann Clark
On 4/9/2010 12:08 AM, Jose Flores Galicia wrote:
Hi.
On the Spa 3102 is set as Dialplan s0:8028 on PSTN line tab, since
other way the incoming call will try to be routed to a non set
extension on [gw8028
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