Something you may want to try (its fixed it for us) is putting an I
(uppercase I) on the asterisk invocation line.
We run servers in the cloud and can't get reliable timing from ISDN
cards etc so this instructs asterisk to generate its own internal
timing. If you have ISDN you probably don't want
SMS when any alarm goes off.
My basic alarmreceiver scripts are available at
http://kevin.withnall.com/2007/07/09/asterisk-alarm-receiver-using-trigg
ers-mysql5/ if anyone wants them.
--
Kevin Withnall http://kevin.withnall.com/
ILB Computing http://www.ilb.com.au http://www.ilb.com.au/
PH: 02
or pointing in the right direction.
Thanks.
--
Kevin Withnall http://kevin.withnall.com/
ILB Computing http://www.ilb.com.au
PH: 02 4227 0001 Mobile: 0412 453 846 FAX: 02 4227 0081
Please consider the environment before printing this e-mail
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as they are differing lengths.
Regards
Kevin
--
Kevin Withnall http://kevin.withnall.com/
ILB Computing http://www.ilb.com.au
PH: 02 4227 0001 Mobile: 0412 453 846 FAX: 02 4227 0081
Please consider the environment before printing this e-mail
-Original Message-
From: [EMAIL PROTECTED
Using trixbox (or a custom dialplan if needed) has anyone been able to
convert a number dialled like
+61242110 to something like 02422110 ie (remove the +61 and
replace with 0)
i just dont know how to set it up, there seems to be no dialplan
wildcard i can use to match +.
I was
etc and they all cal call the Polycom
without problem.
Does anyone know what could be going on ?
Thanks.
--
Kevin Withnall http://kevin.withnall.com/
ILB Computing http://www.ilb.com.au
PH: 02 4227 0001 Mobile: 0412 453 846 FAX: 02 4227 0081
Please consider the environment before printing this e
I have a trunk setup to broadvoice and it seems to work fine. I needed
an indial in another localtion so setup a second trunk on a new bv
account.
Either account regiteres by itself but when both are enabled, it sits in
Request Sent status on sip show registry
anyone else had this issue ? I
been working beautifully.
Run a hdparm /dev/hda (or whatever your disk is) and make sure its in
dma mode.
--
Kevin Withnall
ILB Computing
PH: 02 4227 0001 Mobile: 0412 453 846 FAX: 02 4253 0001
http://www.ilb.com.au/ http://kevin.withnall.com/
-Original Message-
From: [EMAIL
Thers a box we can get here in Australia that has an RJ45 plug, a built
in web server that has a config page and URL's to close/open one of 4
included relays. I use phpagi to hit the url and open/close doors that
way. If anyone is interested, ill let you know URL's but its being sold
from our
I have a patton 1400 setup to handle the bri interface. As a
trixbox user, I wanted a sip trunk rather than having to re-compile bri support
into trixbix.
Anyway, I have it working now so that asterisk can make
calls and they are passed properly to the telephone network. Incoming calls
After much playing and getting nowhere, I was on the phone to the guys
from www.voipshop.com.au and mentioned that the pri dropout problem was
occuring and if they had any solutions.
Immediately they mentioned something that causes a problem in australia.
On longdistance phone calls (sometimes)
Ill actually be working on this shortly. Ive already designed the system
in my head and just need to write it now :-)
The problem is it will be fairly integrated into the alarm response
code.
I don't think it would be hard to write a phpagi script to do this
normally.
It would just have to...
It consists of a MySql database using triggers, and a PHPAGI
script that does the calling.
http://www.voip-info.org/wiki/view/MySql+trigger+based+alarm+response+system+for+AlarmReceiver%28%29
Any comments or fixes are welcome.
Ill work on
a web setup front end so people can
Call Ref: len= 2 (reference 207/0xCF) (Terminator)
---
Call Ref: len= 2 (reference 209/0xD1) (Terminator)
132c108
Call Ref: len= 2 (reference 207/0xCF) (Originator)
---
Call Ref: len= 2 (reference 209/0xD1) (Originator)
--
Kevin Withnall
ILB Computing
PH: 02 4227 0001 Mobile: 0412 453 846
] On Behalf Of Kevin Withnall
Sent: Saturday, 15 July 2006 11:05
AM
To:
asterisk-users@lists.digium.com
Subject: [asterisk-users] PRI
dropouts
Recently we cut over to using asterisk (trixbox 1.1.1) for
our production system.
We are using aTE110P digium card (Primary rate) with a
Telstra onramp 10
Recently we cut over to using
asterisk (trixbox 1.1.1) for our production system.
We are
using aTE110P digium card (Primary
rate) with a Telstra onramp 10.
Sometimes when people call, on their
end it doesnt seem to connect. On our end, we get caller id, it passes ok to
the sip phone but
when to call out ?
Thanks.
--
Kevin Withnall
ILB Computing
PH: 02 4227 0001 Mobile: 0412 453 846
FAX: 02 4227 0081
http://kevin.withnall.com/
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I know its available online as well but I can't get access to it (they
are ignoring my emails).
Can someone please post a sample XML file for the SPA3000 (and the PAP2T
if anyone has one ?)
Thanks for your time.
--
Kevin Withnall
ILB Computing
PH: 02 4227 0001 Mobile: 0412 453 846
FAX: 02 4227
Currently we have (with our NEC phone system) the options in voicemail to
have a message say press 2 to go to my mobile phone
Can this be done in asterisk without setting up an IVR for each user ?
Has anyone got a voicemail dialplan that can do this ?
Thanks
--
Kevin Withnall
ILB Computing
Thanks for that, it works like a charm :-)
--
Kevin Withnall
ILB Computing
PH: 02 4227 0001 Mobile: 0412 453 846
FAX: 02 4227 0081
http://kevin.withnall.com/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Paul Hales
Sent: Tuesday, 4 July 2006
Im trying to use a MeetMe room to record a podcast. The
quality of the wav is rather poor although all parties entering the room are
coming in via PRI channels. It sounds fine while listening in the room but the
recording is very poor.
Im using lame h in.wav out.mp3 to convert it. Any
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