was able to properly execute the command). If your script
doesn't properly handle these responses, you get the error mentioned
below.
It's never caused any of my calls to drop, though. Try turning on AGI
debug to see if this is the case for you.
Thanks,
--Warren Selby
On Jan 26, 2010, at 5:11
are available for the Queue() command in the dialplan.
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the line, as well as allow you to follow several of the
simple how-to guides out there.
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set.
It's just not the easiest path to take, and not necessarily the path the OP
should go down unless he's looking for a challenge.
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http://www.selbytech.com
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as intended.
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Instead of host=dynamic, use host=1.1.1.1, or
host=1.1.1.0/255.255.255.0.
Thanks,
--Warren Selby
On Jan 12, 2010, at 11:16 AM, Aggio Alberto
alberto.ag...@loquendo.com wrote:
Hi guys,
I recently faced an issue regarding SIP registration: I have a 2-NIC
Linux PC, with eth0 set
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are using, you can set the port
that asterisk binds to using the following commands in sip.conf:
1.6.x:
udpbindaddr = x.x.x.x:5061
1.4.x:
bindport = 5061
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'.
Please run ./configure.
make: *** [makeopts] Error 1
And did you run ./configure like the error message says?
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And what is the output of the ./configure? Does it generate any errors?
Thanks,
--Warren Selby
On Dec 22, 2009, at 1:09 AM, hadi motamedi motamed...@gmail.com wrote:
On Tue, Dec 22, 2009 at 6:56 AM, Warren Selby
wcse...@selbytech.com wrote:
On Mon, Dec 21, 2009 at 11:12 PM, hadi
Is the new Fax For Asterisk being released in conjunction with this
release?
Thanks,
--Warren Selby
On Dec 18, 2009, at 4:59 PM, Asterisk Development Team asteriskt...@digium.com
wrote:
The Asterisk Development Team has announced the release of Asterisk
1.6.1.12.
This release
http://www.digium.com/en/products/software/faxforasterisk.php
Thanks,
--Warren Selby
On Dec 18, 2009, at 7:11 PM, Thomas Perron thomas.per...@gmail.com
wrote:
How does Fax for Asterisk work?
On Fri, Dec 18, 2009 at 7:51 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
Warren Selby
the newest FFA modules will be released, or
if there is any way I can help to test the new modules?
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Take the whitespace out of your ()'s. It's:
exten = 80,n,BackGround(es/good)
not
exten = 80,n,BackGround( es/good )
Thanks,
--Warren Selby
On Dec 12, 2009, at 9:16 PM, Landy Landy landysacco...@yahoo.com
wrote:
Same thing:
== Using SIP RTP CoS mark 5
-- Executing [...@outbound:1
Can I install my free fax for asterisk license on more than one
machine? I.e using my digiun account to download the free FFA module,
am I restricted to just the first machine I put it on, or can I put
the free FFA on multiple servers?
Thanks,
--Warren Selby
On Fri, Dec 11, 2009 at 10:30 AM, Leif Madsen
leif.mad...@asteriskdocs.orgwrote:
Warren Selby wrote:
Can I install my free fax for asterisk license on more than one
machine? I.e using my digiun account to download the free FFA module,
am I restricted to just the first machine I put
:
No such command 'restart gracefully' (type 'help restart gracefully' for
other possible commands)
Can anyone think of why this is happening?
Thanks
Maybe you need to escape your quotes (\restart gracefully\) in your
script?
Just a thought...
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Thanks,
--Warren Selby
http://www.selbytech.com
Why not do something with Background()? i.e
Playback(you-have)
SayNumber(${numMessages})
Playback(messages)
Background(press-1-or-2)
Then just be sure to record the audio files in the appropriate
directory...
Thanks,
--Warren Selby
On Dec 3, 2009, at 12:39 AM, Olivier oza-4
You need to purchase a smartnet license for the phone in question in
order to legally get the sip firmware.
Thanks,
--Warren Selby
On Dec 2, 2009, at 11:28 PM, Ricardo Melendez
rmelen...@utep.com.mx wrote:
Hi to All, I am trying to flash to SIP image one Cisco 7941 IP Phone
to work
an error about not being able to find the config file and then the
phone will not boot up. Has anyone seen anything like this before?
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--Warren Selby
http://www.selbytech.com
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Do you have *11 registered in your voicemail.conf file? What does the
cli output look like when you try to leave a voicemail?
Thanks,
--Warren Selby
On Nov 28, 2009, at 7:22 PM, matthieu Nicaise techni...@thinkrosystem.com
wrote:
Hello everybody,
I'm using Asterisk ( 1.6.1.9
19 09 55
techni...@thinkrosystem.com commerc...@thinkrosystem.com
Thinkro System
http://www.thinkrosystem.com/
What is the output of 'ls -lh /var/spool/asterisk/voicemail/default/*11/' ?
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--Warren Selby
http
, since it's not appearing
your cli output. Make sure your extensions.conf file has been saved and
then try dialplan reload in the cli and then try calling extension *11
again.
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is the phone support.
Thanks,
--Warren Selby
On Nov 24, 2009, at 2:49 AM, Olivier oza-4...@myamail.com wrote:
Hello,
LLDP is more and more available on various network elements
(endpoint, switches, ...).
It seems to ease network configuration.
Do you have any experience with it ?
How
There could be many reasons for this. You should show us the output of
your asterisk cli during a failed call attempt, and we can go from
there.
Thanks,
--Warren Selby
On Nov 20, 2009, at 5:23 PM, Brad Darr bd...@juniper.net wrote:
Hello,
I have been working on getting a Cisco 7961G
insecure=very
disallow=all
allow=ulaw
allow=alaw
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any typographical errors (which I mentioned what you
posted contained).
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What does your provider see when you attempt to call them?
Thanks,
--Warren Selby
On Nov 18, 2009, at 3:38 PM, Landy Landy landysacco...@yahoo.com
wrote:
Thanks for replying.
But how come I'm able to use a softphone to place calls from withing
the lan? I really dont get it. What ports
CLI output of calls that go through the local channel instead of the defined
channel would be useful to help diagnose what's going on here.
Thanks,
--Warren Selby
On Mon, Nov 16, 2009 at 4:01 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
I've been noticing an odd issue with our servers
voipprovider)?
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is though...
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Which models of cisco phones (i.e 79x0, 79x1, 79x2, etc). And what do you
mean by VLAN issue.
Thanks,
--Warren Selby
On Sun, Nov 15, 2009 at 7:41 PM, Leif Madsen
leif.mad...@asteriskdocs.orgwrote:
Julian Lyndon-Smith wrote:
We have several types of phones, Cisco 79xx, Aastra 9133i etc. We
already
linked to them in another post to the list).
Thanks,
--Warren Selby
On Thu, Nov 12, 2009 at 4:11 PM, Stephen Reese rsre...@gmail.com wrote:
On Wed, Nov 11, 2009 at 9:34 PM, Warren Selby wcse...@selbytech.com
wrote:
The 7960 and 79x2 use different sip firmwares and as far a I have seen
://www.selbytech.com/2009/10/setup-cisco-7941-or-7961-with-asterisk/
In that post is a sanitized version of my conf file that I use on my
own deskphone, if you'd like to download it and try it out with your
setup.
Thanks,
--Warren Selby
On Nov 10, 2009, at 6:32 PM, Stephen Reese rsre...@gmail.com
,
--Warren Selby
On Nov 9, 2009, at 8:35 PM, Stephen Reese rsre...@gmail.com wrote:
On Sat, Nov 7, 2009 at 11:36 AM, Warren Selby
wcse...@selbytech.com wrote:
I think your featureLabel definition is wrong.
On the login issue, ssh to the ip of the phone and login first with
the user/pass you defined
,
--Warren Selby
On Nov 7, 2009, at 9:45 AM, Stephen Reese rsre...@gmail.com wrote:
On Sat, Nov 7, 2009 at 12:56 AM, Warren Selby
wcse...@selbytech.com wrote:
That typically means you've got an error in your phone specific
config file,
the SEP[MAC].cnf.xml.
You need to login to the phone via
Probably after 1.6.2 has been officially released beyond the release
candidate stage.
Thanks,
--Warren Selby
On Mon, Nov 2, 2009 at 4:14 AM, Khaled W Chehab kche...@xplorium.comwrote:
When we can expect to have a res_fax and res_fax_degium module for
asterisk V 1.6.2
Regards
What version of asterisk are you installing?
Thanks,
--Warren Selby
On Mon, Nov 2, 2009 at 5:59 AM, Dan Journo d...@keshercommunications.comwrote:
Hello,
Does anyone know where I can get an up to date guide on installing
CDR_MSQL?
VOIP-Info has old information.
Many thanks
Dan
rough idea at the moment, but hopefully it gives you some ideas to work
with.
Thanks,
--Warren Selby
On Fri, Oct 30, 2009 at 12:35 PM, Danny Nicholas da...@debsinc.com wrote:
Have you tried “forward-porting” it? I don’t do queues or 1.6 so it’s
just an academic question to me
How are you setting up xlite and the ata? Which version of Asterisk are you
using? What does the general section of your sip.conf look like?
On Fri, Oct 30, 2009 at 1:01 PM, Cliconnect cliconn...@cliconnect.comwrote:
Hi all,
I can only get a line signal when I set the phones to not
You're attempting to connect on ports 5061-5062 but are bound to port
5060...?
What does your CLI look like during a failed call attempt?
Thanks,
--Warren Selby
On Fri, Oct 30, 2009 at 2:18 PM, Cliconnect cliconn...@cliconnect.comwrote:
Thank you,
How are you setting up xlite
of the atftpd instances)
/home/phones/cisco/7961 (root directory for another atftpd instance)
Then, with a little dhcpd.conf magic, you can easily point different sets of
phones to different tftp servers, using pools that match on key Product ID
(I think) strings.
--Warren Selby
On Fri, Oct 23, 2009
Have you tried accessing the IP address of your server from another
computer's web browser?
--Warren Selby
On Fri, Oct 23, 2009 at 10:19 AM, giancarlo lombardo
gianclomba...@gmail.com wrote:
Dear all,
I just installed asterixnow,
but no graphical interface start automaticaly neither linux
Have a quick look at this guide on NAT and SIP -
http://www.aocomputing.net/?p=3. This is the link given if you were to ask
this same question in the IRC channel...
--wcs
On Wed, Oct 21, 2009 at 9:59 AM, Ott Rose sixfourimp...@hotmail.com wrote:
Here is what i think the is helpful from
vitelity.
Thanks,
--Warren Selby
On Oct 9, 2009, at 4:48 PM, Ken D'Ambrosio k...@jots.org wrote:
Hi, all. I'm probably doing Something Dumb(tm), so please feel free
to
point out whatever I'm missing, no matter how stupid.
Anyway, I've got IAX set up to Vitelity. When I try to call my DID
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