)
** **
Try changing the Page() to a Dial() command and see if that makes a
difference.
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--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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Install OpenSSL-devel (or whatever the equivalent ubuntu package is called) and
then recompile / reinstall and test it again.
Thanks,
--Warren Selby, dCAP
On Aug 2, 2011, at 12:06 PM, neo haux neo.h...@gmx.com wrote:
Hi,
I´ve compiled asterisk-1.8.5.0 on my Debian based distro (Pinguy
to 10.0 is just
confusing.
[1] -
http://lists.digium.com/pipermail/asterisk-announce/2011-July/000331.html
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accordingly? Is
that done in sip.conf? or extensions.conf?
Usually this is handled on the phone itself or within one of it's
configuration files.
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--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
Looks like you need an 's' exten in your [internal] context.
Thanks,
--Warren Selby, dCAP
On Jul 13, 2011, at 2:02 AM, Malvin Rito mr...@mail.altcladding.com.ph wrote:
Hi List,
I have another issue on allowing outgoing calls to PSTN on Asterisk via Avaya
Phones, I hope that anyone could
?
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On Fri, Jul 8, 2011 at 9:00 PM, Doug Lytle supp...@drdos.info wrote:
Warren Selby wrote:
Not trying to start a war here,
That may be, but I have experience with VB.
Doug
I use VB on my main desktop that runs windows in order to setup test
environments, so I understand. :)
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do. Could this be the cause? Dahdi_test
shows 99.9xxx% accuracy with the dummy timer.
** **
Regards,
** **
** **
Mike
** **
I had this happen on a client's server where the HDD was failingmaybe
integrity check the disk?
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http
Pretty sure with Polycom's you can only specify in sequential order, you can't
pick and choose the buttons you assign.
Thanks,
--Warren Selby, dCAP
On Jun 28, 2011, at 11:58 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
I've got a Polycom SPIP 650 3.3.1F-enabled phone with an 14-keys side-car
issues with your Dial statements, but I'm on my
phone right now and can't really diagnose them. I'll take a look later when I'm
back at a desk, if no one else has commented by then.
Thanks,
--Warren Selby, dCAP
On Jun 28, 2011, at 12:30 PM, motty.cruz motty.c...@gmail.com wrote:
Hello, I
be a little sarcastic... :)
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properly with all of the appropriate
config files, plug your phone in and follow the instructions at the bottom
part of my blog post that explain how to get the phone reflashed to the SIP
image and registered to your asterisk server.
--
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--Warren Selby, dCAP
http://www.SelbyTech.com http
On Mon, Jun 20, 2011 at 5:38 AM, bilal ghayyad bilmar...@yahoo.com wrote:
Dears;
snip
Have you thought about perhaps just flashing the phones to use the SIP
firmware?
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easier (and safer!) if people
posted simple responses on this list when suggestions worked for them...
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, if you're not getting certain variables to pass into the 'h' extension,
that you feel should indeed be passed into the 'h' extension, that may be
considered a bug...but you would need to show us CLI output and existing
dialplan for followup.
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--Warren Selby, dCAP
http://www.SelbyTech.com http
playing because it has no
additional steps and nothing that will tell it to go to the 't' extension.
Also, consider switching your dialplan priorities away from 1,2,3... and
go to 1,n,n,n... as this reduces headaches in the longrun.
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for your example it
would be
${EXTEN:1:1}
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-through to a hylafax server.
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on there that
details how to setup a 79x1 phone using SIP firmware with asterisk. Click
the link in my signature and go to the Blog and you should be able to easily
find the relevant post.
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http://www.SelbyTech.com http://www.selbytech.com
. You should see SOMETHING on the CLI during the call. Post that
output to the list and we can help you from there.
This does not always indicate someone attempting to hack you, I've seen this
occur when there are line errors on FXO devices (among other things).
--
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settings and try again. A SIP debug trace would be very
useful for debugging this (sip set debug on on the asterisk CLI or tcpdump
-l -n -s 0 -w sipdebug.pcap port 5060 from the command line).
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), and associated those with gchat accounts (
wcse...@selbytech.com), and successfully received calls on my asterisk using
this solution.
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I have a free google apps account (http://www.google.com/a I think) setup for
SelbyTech.com. Basically it is a gmail account, just with a different domain.
Thanks,
--Warren Selby, dCAP
On Jun 17, 2011, at 2:43 AM, asterisk asterisk aster...@ck-lee.com wrote:
Could you elaborate on how you can
process. Another option is
the externnotify= command, but that is run on more occasions than just when
a voicemail is left.
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of did's
assigned to them but only a small number of extensions that need direct dial
capabilities.
Thanks,
--Warren Selby, dCAP
On Jun 3, 2011, at 2:34 PM, Jesse Thompson jes...@gmail.com wrote:
(reposted with correct subject line, I think messing up the subject
line last time prevented my
(5);
copy(/var/spool/asterisk/tmp/$callerid-$timestamp.call,
/var/spool/asterisk/outgoing/$callerid-$timestamp.call) or die copy
failed: $!;
exit;
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--Warren Selby, dCAP
Our website just got a facelift! Check it out!
http://www.SelbyTech.com http://www.selbytech.com
Block inbound udp port 5060 using your firewall?
Thanks,
--Warren Selby, dCAP
On May 27, 2011, at 10:45 AM, vip killa vipki...@gmail.com wrote:
Is there a way to disable all SIP registration and block any requests? The
reason I'm asking is this particular Asterisk server will just
/msg207540.html
A quick google for magicjack quality C. Savinovich turned this up as the
second hit...that being said, I agree the OP from today is just spam and
doesn't really help anyone.
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Our website just got a facelift! Check it out!
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you should
be able to setup a SIP trunk. I've been able to successfully integrate a
Cisco CallManager 7.x system with Asterisk using SIP trunking, so I imagine
you should be able to do the same here.
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--Warren Selby, dCAP
http://www.selbytech.com
Show us the cli trace of the delay.
Thanks,
--Warren Selby, dCAP
On May 10, 2011, at 2:18 AM, Pezhman Lali l...@lopl.net wrote:
thanks,
this delay is occurred on asterisk server, between dial execution and
CALLED .
On Mon, May 9, 2011 at 7:12 PM, Warren Selby wcse
- I'll just be along for the
ride then :D
Sebastian
When I used to have a Cisco 7941 phone in my home office, I only needed the
tftp server online if I made changes to the configuration. The phone itself
worked fine without one being online all the time.
--
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http
on your phone itself. What model phone do you have?
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differently?
Sebastian
All I can really think is that the 7940 and the 7941 use different firmwares
and configuration files, maybe there was some kind of change between the
two? Does the phone never time out while looking for a server?
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http://www.selbytech.com
the filename for the recorded call using Set(MONITOR_FILENAME=blah)
before you call the Queue() command.
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.
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.
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Set a variable ${_CALLED_EXT} to ${EXTEN} before you hang up the call, then
reference that variable in your h exten.
Thanks,
--Warren Selby, dCAP
On May 5, 2011, at 11:59 AM, satish patel satish...@hotmail.com wrote:
Hi All,
I am using
http://www.theschmandts.org/blog/2007/05/05/email
I forget the term, but basically the variables you set on a current active
channel are only accessible on that channel. In this case the variables are
specific to the specific call in progress.
Thanks,
--Warren Selby, dCAP
On May 5, 2011, at 1:02 PM, satish patel satish...@hotmail.com wrote
And Sherwood beats me to the punch again :).
Thanks,
--Warren Selby, dCAP
On May 5, 2011, at 1:15 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote:
No, the variables are channel specific except for when they're inherited,
which doesn't affect you here
On Thu, May 5, 2011 at 1:02 PM
PQMSTATUS is set only after you run the application PauseQueueMember().
Thanks,
--Warren Selby, dCAP
On May 5, 2011, at 2:11 PM, Louis Carreiro carreir...@gmail.com wrote:
Hey all!
I'm trying to do a bit of logic here so that a user only has to dial one code
to pause/unpause in a queue
by instructing them
to dial one key code to pause or unpause.
v/r,
Me
On Thu, May 5, 2011 at 6:04 PM, Warren Selby wcse...@selbytech.com wrote:
PQMSTATUS is set only after you run the application PauseQueueMember().
Thanks,
--Warren Selby, dCAP
On May 5, 2011, at 2:11 PM, Louis
, RQMSTATUS) to
either ADDED or MEMBERALREADY.
I still think I managed to get the Pause and UnPause functionality on one
button, but I need to check the actual code I used, and I can't do that
until tomorrow. Maybe someone else will have chimed in by then...
--
Thanks,
--Warren Selby, dCAP
http
/${EXTEN}@POTS1,60,o)
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You're using 1.4.2. Why not try upgrading to a more recent release of 1.4 (I
believe 1.4.41 is current) and see if your issue has been resolved.
Thanks,
--Warren Selby, dCAP
On Apr 29, 2011, at 7:32 AM, Rizwan Hisham rizwanhas...@gmail.com wrote:
Yes I have it there, here the content
Which version of 1.4 is current, 1.4.41 or 1.4.40.2? I received both just now
and actually received the notification for 1.4.40.2 AFTER I got the one for
1.4.41...
Thanks,
--Warren Selby, dCAP
On Apr 26, 2011, at 12:01 PM, Asterisk Development Team
asteriskt...@digium.com wrote
On Mon, Apr 18, 2011 at 4:54 AM, Jonas Kellens jonas.kell...@telenet.bewrote:
Hello list,
I have in sip.conf :
snip
So are my settings wrong ?
What does sip show settings look like from the CLI?
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
. Try running a packet capture using tcpdump to see if your
asterisk box is getting any traffic from the phone, etc. Basic network
troubleshooting at this point. Can you ping the box from your network, can
you ping the phone from your box, etc?
--
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http
Your last line in the dialplan should be StartMusicOnHold(), not just
MusicOnHold().
Thanks,
--Warren Selby, dCAP
On Apr 11, 2011, at 6:24 AM, virendra bhati virbh...@gmail.com wrote:
I am using Elastix. Asterisk is used for PBX application in Elastix. I want
to make customize MOH
Doesn't Elastix have it's own tool for MusicOnHold? Maybe check with that and
see if that makes a difference.
Thanks,
--Warren Selby, dCAP
On Apr 11, 2011, at 12:49 PM, virendra bhati virbh...@gmail.com wrote:
Yes ,
I show me the all configured MOH. But don't play the MOH.
After 12
Add exten = 6000,n,StartMusicOnHold() to the end of your current dialplan and
try again.
Thanks,
--Warren Selby, dCAP
On Apr 8, 2011, at 1:51 AM, virendra bhati virbh...@gmail.com wrote:
I am using Elastix. Asterisk is used for PBX application in Elastix. I want
to make customize MOH
something simple.
DYNAMIC_FEATURES=MixMonApp, either declared in your globals section of
extensions.conf, or used in a Set(DYNAMIC_FEATURES=MixMonApp) fashion on a
per channel basis in extensions.conf.
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http://www.selbytech.com
from there. My guess is you're going to at the least get the
preconfigured email address and the contents of your emailsubject and
emailbody options (both of which have the option of passing multiple useful
variables).
--
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--Warren Selby, dCAP
http://www.selbytech.com
/7623
[Apr 5 13:51:26] WARNING[9539]:
/usr/local/src/asterisk/asterisk-1.8.3.2/include/asterisk/crypto.h:145
__stub__ast_aes_set_encrypt_key: AES encryption disabled. Install OpenSSL.
Do you have OpenSSL installed?
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
. And for some reason people seem to think
that it requiring Python is a bad thing. But then again, I'm not running it
on small systems - most of the systems I've put it on have plenty of excess
cpu and memory, so that hasn't been an issue for me.
--
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http
output, from beginning to end of each call. With this kind of
information we can begin to diagnose what's happening.
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for you? Or perhaps you're just out looking to troll?
From the page you linked:
1. Responses should be placed under the original quoted text.
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The Polycom 501 has basically been replaced by the Polycom 550.
Thanks,
--Warren Selby, dCAP
On Apr 1, 2011, at 4:25 PM, satish patel satish...@hotmail.com wrote:
We're looking to purchase new phones for Asterisk. There are a limited
number of new Polycom 501's on the market, mostly
. Thanks for the quick fix Tilghman!
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say,
Polycom or Aastea phones. But no matter, whichever phone you chose, you'll
likely have to do any custom button assignments in the phone's config, whether
that be a file or a webapp.
Thanks,
--Warren Selby, dCAP
On Mar 30, 2011, at 6:10 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Kindly
)}) *
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troubleshoot your exact issue.
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, does it properly create the SIP packets? For some reason, I'm
thinking this is just the way it is, but someone closer to the the actual
sip development may be able to better tell you. Perhaps open a ticket on
the bug tracker?
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get
the same issue. If I do, I'll open a new bug on the issue tracker, and I'll
post my results here on the list.
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chan_dahdi.conf file (I believe), probably using groups (although I
suppose you could just use channel numbers).
I believe you can verify that libpri has been compiled into asterisk by issuing
a 'pri show' command at the console after you've got everything running.
Thanks,
--Warren Selby
The answer to all of your questions are the same - the config file that you
create for your phone.
Thanks,
--Warren Selby, dCAP
On Mar 29, 2011, at 5:16 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hello;
I need to use Cisco IP Phones with Asterisk and I have some questions to know
how
On Tue, Mar 29, 2011 at 4:03 PM, Warren Selby wcse...@selbytech.com wrote:
That information does indeed look like what I want and it appears to be
setup correctly. I will be building a comparable test system later today
(using all the same software versions as you) and I'll test to see if I
]: http://www.voip-info.org/wiki/view/Asterisk+cmd+GotoIfTime
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in sip.conf.
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.
No, I don't believe so, but the best way to find out is to test.
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On Wed, Mar 16, 2011 at 11:41 PM, Warren Selby wcse...@selbytech.comwrote:
On Wed, Mar 16, 2011 at 11:39 PM, edward choi mp2...@gmail.com wrote:
Thanks for the info.
But then do I have to set 'nat=no' when he is on a public IP address?
It would be quite a labor to switch back and forth every
the custid variable, and I'm pretty sure
there is also a QUEUEMEMBER variable that's set with the agent extension (not
sure of the variable name, that's just off the top of my head).
Thanks,
--Warren Selby, dCAP
On Mar 10, 2011, at 3:37 PM, Danny Nicholas da...@debsinc.com wrote:
From
On Thu, Mar 10, 2011 at 4:28 PM, Warren Selby wcse...@selbytech.com wrote:
In your AGI you should be able to read the custid variable, and I'm pretty
sure there is also a QUEUEMEMBER variable that's set with the agent
extension (not sure of the variable name, that's just off the top of my
(or whatever is the actual location of your
mysql.sock file).
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or even on
systemm/...
knowing the server performance only the software side includes any cpu
history like when the server is busy or idle
Have a look at munin, or maybe cacti or even mrtg.
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
Pretty sure I saw those on wiki.asterisk.org.
Thanks,
--Warren Selby, dCAP
On Feb 28, 2011, at 10:39 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
Hi,
Would someone know where I can download the CEL schema for (create commands)
for PostgreSQL please ?
--
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of the call doesn't
offer SRTP, then there won't be any SRTP in the call path.
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the service again
from the command line.
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(num)} variable before you make your outbound call.
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On Thu, Feb 24, 2011 at 5:01 AM, Israel Gottlieb isr...@gmail.com wrote:
sorry i wasnt clear enough i meen inbound
You could always Answer() the call in your dialplan before you do anything
else, then Dial() whoever you're trying to reach and set your own timeouts
there.
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actually happening.
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= _4XXX,1,Verbose(Calling roaming extension ${EXTEN})
exten = _4XXX,n,Set(ROAMEXT=${DB(roam/${EXTEN:1})})
exten = _4XXX,n,Dial(SIP/${ROAMEXT},30)
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On Thu, Feb 24, 2011 at 10:32 AM, Jesse Cloutier je...@cronomagic.comwrote:
Whats the best way to start tracking this down?
Collect proper debug information[1].
[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
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http
.
Without more details, I'm not sure how much help you're going to get. Show us
some console output of the issue, capture the proper debug logs, etc, and
perhaps you'll find more help.
Thanks,
--Warren Selby, dCAP
On Feb 23, 2011, at 11:57 AM, vip killa vipki...@gmail.com wrote:
I'm sorry i
a new issue on the bug tracker and it will get looked at. It's
a pretty painless procedure.
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New
You're not properly reading in the response after each NoOp you send out. Each
time you send something to asterisk in AGI, you must read the response in your
script.
Thanks,
--Warren Selby, dCAP
On Feb 22, 2011, at 4:39 AM, Gilles codecompl...@free.fr wrote:
Hello
Incoming calls from
match, I'm pretty sure there is a variable you can use to
achieve the same effect.
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New to Asterisk
On Mon, Feb 21, 2011 at 12:37 PM, Warren Selby wcse...@selbytech.comwrote:
Then you need to include the [roaming-ext] context in whatever context your
phones dial from. The basic idea behind this is that you need to store the
extension where your roamer is currently sitting in your DB, which
It's been my experience that the MEMBER... Variables are populated by the
person who answers the queue call. If no one answers the call, I would imagine
the variables would be null.
Thanks,
--Warren Selby, dCAP
On Feb 20, 2011, at 2:17 AM, magnu...@inputinterior.se wrote:
Hmm,
First i must
I'd love to have a look at this also. You may want to consider adding it to
one of the wiki's out there, as well.
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http://www.selbytech.com
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numbers starting with a 6:
exten = _6XXX,1,Verbose(Test1)
exten = _[123457890]XXX,1,Verbose(Test2)
same = 2,Verbose(Test3)
There may even be a better pattern exclusion match parameter...this is just
what I came up with off the top of my head.
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
, too.
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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You've got connection=jp_jabber defined in one file, and [jb_jabber] defined in
the other.
Thanks,
--Warren Selby, dCAP
On Feb 10, 2011, at 5:55 PM, William Stillwell will...@stillwellsoft.com
wrote:
Sorry, Asterisk Build 1.6.2.7
From: asterisk-users-boun...@lists.digium.com
adding the following to your [google-in] context in extension.conf:
exten = _.,1,Verbose(Call from GTalk - catchall)
exten = _.,n,Set(CallerID(Name)=From GoogleTalk)
exten = _.,n,Dial(SIP/1000)
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
(handle-hangup)
Try taking the quotes () out of the line that says Internal call. So it
should be:
exten = _312,1,Set(CALLERID(name)=Internal call)
...and see if that helps.
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
are affected too.
Do you have DAHDI installed and running? Show us the output of dahdi_test
from the command line.
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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.
If that isn't the case - maybe look at using a local channel instead of
SIP/xlite to setup the call?
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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New
,
--Warren Selby, dCAP
http://www.selbytech.com
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Turn up the verbosity on your console, play a sound file then paste the output
in a reply.
Thanks,
--Warren Selby, dCAP
On Jan 28, 2011, at 4:27 AM, Сикорский Сергейs.sikor...@lanet.ua wrote:
Hi.
I've updated sound files in /var/lib/asterisk/sounds and restarted Asterisk,
but it still
switch, you need to stop it using the command core stop now.
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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