Hi,
I have two cisco phones with sip images and I am trying to configure
to work with asterisk. Both can call demo numbers and voicemail etc.
but can't call each other.
sip show registry and sip show users both indicate that asterisk
doesn't know the phones ip addresses, and when u try to
Hi,
I am using a sip provider that offers voicemail. They send me a sip
notify that there are voicemails, and I would like this notify to be
sent to one of the extensions on asterisk (a sipura 2100 or cisco
7960), to light a lamp/give stutter dial tone.
The provider is running * too and is
Hey Stefan,
Have you had a look at http://www.voip-info.org/wiki/view/Asterisk+cdr+odbc?
Voip-info.org isn't official, but it's a pretty good site.
Have a look under Setting up the CDR Database/Table.
There's also the AST:
https://wiki.asterisk.org/wiki/display/AST/MSSQL+CDR+Backend
Thanks very much for that mate. Fantastic guide - I'll look at implementing it
on our services some time soon!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Pocock
Sent: Monday, 20 August 2012 12:49
question, AST = asterisk or is it by chance just a part of asterisk or
sth. like this?
2012/8/20 Andrew White
and...@computersforall.com.aumailto:and...@computersforall.com.au
Hey Stefan,
Have you had a look at http://www.voip-info.org/wiki/view/Asterisk+cdr+odbc?
Voip-info.org isn't official, but it's
The built in file convert function has saved my bacon when I need to convert
quickly.
Apart from that, there are a ton of examples with external programs on
voip-info.org:
http://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk
I usually custom code it. I've got a PHP interface set up on one of our
asterisk boxes (locked down via IP range) that allows calls to be initiated via
the script. I can interface with this via standard HTTP requests from anything
I want. I've written solutions as Chrome/Firefox addons, local
Hey Brian,
I don't know exactly how to find this out (although I'd love to!) but I know it
is possible. I saw a video on a custom web socket based AMI connection that
showed when the channel was on hold.
I'd assume if you open an AMI interface, log all of the calls and ensure you
put a
Is realtime an option for you to install?
If it is you could develop a web interface that allows you to put the customer
account number in, or even integrate it into your existing customer management
system. Depending on the scale of what you're doing though, this might be
overkill.
Hey Motty,
The simplest way I've found is having an asterisk console open (asterisk -r)
with verbosity to level 12. Alternatively you could tail -f the full log (in
/var/log/asterisk) - I like to parse it with something like ccze to colour code
things.
The better solution I've found is to use
Hey Dan,
Please keep us updated on a video or transcript of this talk - this seems like
a very fascinating presentation and I'd love to get more information.
Cheers,
Andrew.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan
Hey Longst,
I'd recommend having a look into the LUA support Asterisk offers for its
dialplans or AGI. These are the only realistic ways to add functions, unless
you want to write your own C module and compile it in. Adhearsion is an option
as well, if you are proficient with ruby.
Cheers,
Hi Dmitry,
I've been using the below for a long time and it works perfectly. I've built a
custom solution in terms of the tables.
My understand of the external queue logging to MySQL is that it uses the
Realtime engine to do so, so it's not affected by app_mysql directly. Please do
correct me
Hey all,
I've built a custom application for our call center and am having one problem.
Unfortunately certain things happen whilst the agent has the customer on hold
which I'd like to work around. But I can't work out how to catch the actual
hold event so I can do something about it. From the
Hey guys,
I've got a part of my dialplan that dials multiple people:
exten = direct,n,Dial(${QUEUEEXTS},${RINGTIME})
Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g.
SIP/100SIP/101SIP/105 etc
This works great, however I want to see if I can find a way to work out
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialplan - working out when users answer
On Wed, Dec 19, 2012 at 10:53 AM, Andrew White
and...@computersforall.com.aumailto:and...@computersforall.com.au wrote:
Hey guys,
I've got a part of my dialplan that dials multiple
to reach a non-existent
destination for gosub: (Context:queueControl, Extension:s, Priority:1)
I can't quite work out why it would be trying to s/1 instead of
direct/CONNECTED =/.
Any ideas?
Thanks!
From: Andrew White
Sent: Wednesday, 19 December 2012 5:58 PM
To: Asterisk Users Mailing List
-Commercial Discussion
Subject: Re: [asterisk-users] Dialplan - working out when users answer
On Wed, Dec 19, 2012 at 12:44 PM, Andrew White
and...@computersforall.com.aumailto:and...@computersforall.com.au wrote:
Hi Satish/list,
Looks like I spoke to soon.
I have the following in my dialplan:
Dial
should have extension s with priority 1 in
this context.
--Satish Barot
On Mon, Jan 7, 2013 at 12:08 PM, Andrew White
and...@computersforall.com.aumailto:and...@computersforall.com.au wrote:
Hi Satish,
Thanks for your response - sorry on the slow reply.
So I've tried the following in the dialplan
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