Displaying 20 results from an estimated 53 matches for "leifmadsen".
2009 Jul 16
5
AGI to announce temperature from weather.com XML file
I would like to have the ability to have Asterisk announce the temperature
-- not using TTS -- within the dialplan.
For a non-Asterisk project, I have a cron job that periodically pulls down
an XML file from weather.com containing local weather data (TWC's user
agreement requires that data be cached locally). Using sed, I also create a
text file that contains only the numeric value of the
2004 Dec 09
2
MeetMe Features
Hi all,
I had a chance to use some call conferences that had some very neat
functionalities:
- When you call you are first asked for your name
- When someone joins the conference a message "<name> is now joining the
conference." is played.
- When someone leaves the room a message "<name> has left to conference." is
played.
How can I set MeetMe/Asterisk to have
2006 Jan 16
5
Dundi Examples
Can someone show me how to set up DUNDi, I will be using it to connect
14 asterisk servers internally. I don't want to use it on the external
world. If anyone has any examples of connecting 2 or 3 (if their is a
difference) machines in a DUNDi co-operation that would be helpful.
Johnathan Falk
Network Administrator
Clinton Community Schools
2005 Oct 15
7
You ASKED for an Asterisk book, you GOT an Asterisk book!
...the number of people accessing it at once, we appoligize and
appreciate your patience. For those of you who are able to obtain the
full copy, please consider helping us out by creating mirrors and
torrents and posting them to the list by replying to this thread.
Thanks!
--
Leif Madsen - http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk
2009 Jun 28
0
Looking for real world uses of CallbackAgentLogin()
...ty.
However, after looking at the AgentCallbackLogin() application, it seems there
is a fair amount of functionality that it could be used in, so I'd like to try
and cover the common, and not so common, methods that it is being used in currently.
If you could reply directly to me at leif at leifmadsen.com with a snippet of your
dialplan and a brief description of what you're doing with AgentCallbackLogin(),
it would be appreciated. I will attempt to cover all the most common uses of
AgentCallbackLogin() based on this feedback, and cover how to convert from using
it to a pure dialplan m...
2009 Jul 22
3
Inquiry abount Asterisk "extensions.conf"
Dear All
Can you please let us know how we can modify our Asterisk "extensions.conf"
file so it interprets the subscriber dialed digits in one-by-one digit
manner . At its current configuration , it interprets them in an whole
packet . I mean , say the subscriber dials as "665 0000" so we need Asterisk
to send it to the peer switch as 6,6,5,0,0,0,0 but not as one
2005 Feb 19
3
simpletelecom.com??? are they a SCAM?
Hi List!
any body use www.simpletelecom.com?
I subscribe to www.simpletelecom.com for A-Z termination and paid
US$15.00 and US$70.00 via credit card in two days, but my account has
US$15.00 only. I checked my credit card from the bank and they said me
the payment already paid to merchant.
I've lost US$70.00 :(
so anyone here has experience with them? are they a SCAM?
Thanks!
</Madhawa>
2009 Aug 21
1
Queue Question
First off this is not my work for extensions.conf it is modified from
http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbackl
ogin-to-standard-dialplan-methods-part-1/
So credit to Leif Madsen <http://www.leifmadsen.com>
But as to my question
[AgentLogin]
;A replaced version of AgentCallbackLogin() using a GoSub()
;
exten => XXXXXXXXXX,1,Ve...
2009 Sep 22
2
Problem with dialplan -> gotoif ?
Hi
This is the output from show dialplan dial-sipmnf-sippt-pstn
[ Context 'dial-sipmnf-sippt-pstn' created by 'pbx_config' ]
's' => 1. Verbose(1,Dialing ${ARG1} on mnf pt pstn) [pbx_config]
2. Dial(SIP/${ARG1}@${SIPMNF},${ARG2},${OUTBDIAL}) [pbx_config]
3. Set(GLOBAL(FOUNDME)=${DIALSTATUS}) [pbx_config]
2005 Feb 08
12
SRV lookups
Hi everyone,
I have a question concerning DNS SRV lookups. The situation is like this:
- one central Asterisk server
- many domains with SRV records, let's say we have bar.com and doe.com
Now the question is: if the SRV lookup is done for foo@bar.com the call is
mapped to foo@myasterisk.mydomain.net. Is that correct?
If so, I have a problem: if somebody calls foo@bar.com, Asterisk
2008 Jun 10
3
Asterisk : using setvar with IP Realtime and variable inheritance
Hi,
I have what I think is a relatively advanced question. Any help is
appreciated, even if it's not a complete answer.
I am using Asterisk in mostly realtime fashion, specifically SIP
registrations are in a MySQL table. This works fine (mostly). I also set a
few variables in the setvar column, like this:
callerid_internal=test <710>;did=5555551234
Again, this works
2004 Dec 29
5
zapata.conf not being parsed by *
I am running * 1.0.3 for some reason when I start * is does not appear to be parsing my zapata.conf file. I do not see any errors * just does not seem to know to look for zapata.conf. I am unable to use my FXO card to make calls or receive calls. I have been able to configure SIP to work correctly.
Any help would be greatly appreciated, I spent most of last night searching for an answer.
2006 Jan 17
6
OT: DCAP Certification
Hi,
emails to astricon.net seems to bounce (at least for me)
I need information about proper & authorized Asterisk training in the
Miami, FL area and the possibility of later DCAP testing.
Thanks,
--
-------------------------------------------
Erick Perez
Linux User 376588
http://counter.li.org/ (Get counted!!!)
Panama, Republic of Panama
2005 Jan 08
4
Toronto?
Anyone in the Toronto area interested in getting together to share notes
and swap war stories?
--
Jim Van Meggelen
jim@vanmeggelen.ca
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.6.9 - Release Date: 06/01/2005
2005 Jan 29
1
FS- ideal starter pack. 1 X100P and 1 Grandstream Budgetone-101
Hi, not sure if it is against the rules to sell second hand equipment in
here but haven't seen anything against it so here it is.
I'm upgrading to 2 lines so I have some spare equipment for sale here.
This is an ideal starter pack and will get you going with 1 line and 1
extension.
1 x http://www.digium.com/downloads/product_sheets/X100P.pdf
1 x
2005 Feb 02
2
Installation on Fedora 3
I'm having problems trying to run zaptel. I don't have the hardware, I
first want to test out asterisk. The problem is the usb-uhci/usb-ohci
module, it isn't present on the system as same as usbcore and I don't
know why. Any tip?
--
-DdC
2005 Feb 08
1
CVS or release?
hi
is the v1-0 CVS branch supposed to be stable as in STABLE, or should
one use releases?
roy
2005 Feb 15
1
asterisk qualified
Good day all
Is there any time of VOIP/SIP/asterisk qualifications or certificates?
Thanks
Altus
2005 Sep 27
2
Review: Digium TE405P v2
Hello,
We have finished our tests of the new Digium firmware on the quad T1
cards(TE405P/TE410P). Overall it is a big improvement over the version 1
firmware.
Here's the review:
http://astguiclient.blogspot.com/2005/09/digium-405p-v2-review.html
MATT---
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2005 Sep 30
1
strange wave like noise on sip handset
Hello
On a Sipura SPA-841 handset (and also at other end) you hear a sea wave like
sound - it gets louder then softer and continually repeats.
I don't remember hearing this when using other handsets. But what is this
effect? How can I reduce it?
Angus