Congrats! Enjoy the time away, you've earned it :)
---
Josh Reynolds
josh@engineered.online
On Wed, Aug 17, 2016 at 8:37 AM, Vincent Medina <vmed...@apcn.net> wrote:
> I just wanted to wish all of you good luck I'm officially retired and will
> be removing my name from the list.
card / port, or a config issue.
-Josh
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Rossi (license 6409) Review:
https://reviewboard.asterisk.org/r/3404/
-Josh
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On Tue, May 27, 2014 at 12:31 PM, Sevana Oy sa...@sevana.fi wrote:
Hi,
How do you figure out if one of gateways in your network leads to voice
quality loss f.e. due to transcoding? The point is that all VoIP metrics in
this case remain the same
Thanks!
Sevana
http://www.sevana.fi
Here are links to the Asterisk Wiki for CDR and SIP tables. I didn't find
extensions listed, but it's pretty simple and I can provide the structure
for that if needed, but it would be without a definitive source beyond me
having used it for years. :-)
It's possible. Might want to look through everything here:
http://www.voip-info.org/wiki/view/SMS
On Fri, May 16, 2014 at 11:08 AM, Jayson Devor jayson.de...@gmail.comwrote:
Hello Everyone,
We have an order for SMS messaging. Can you gents and ladies be kind
enough to
disclose if SMS
on the substring:
exten = _1NXXNXX,1,GotoIf($[${EXTEN:-7:3} =
555]?outbound-411,411,1)
This example would match any 1+Area code+number where the prefix is 555.
You could play with your pattern match to catch call to 1+AC+Number and
just AC+Number using this same test since it's right-delimited.
Josh
On Thu, May 8, 2014 at 4:42 PM, Kevin Larsen
kevin.lar...@pioneerballoon.com wrote:
From: Josh Metzger joshdmetz...@gmail.com
If I recall correctly, the only reason we didn't like the built in paging
feature is that it would put a paging soft button on every phone where we
enabled
with the Multicast address/port displayed? I've also run a
wireshark capture and all I see is the RTP stream from my phone to the
server - nothing going back out. What am I missing, here?
Thanks,
Josh
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On Thu, May 8, 2014 at 10:22 AM, Kevin Larsen
kevin.lar...@pioneerballoon.com wrote:
From: Josh Metzger joshdmetz...@gmail.com
I'm currently working with Asterisk 11.8.1 trying to get Multicast
RTP working (it's not) with some Polycom phones, and I'm really
trying to determine
MACRO_RESULT or GOSUB_RESULT (depending on which you used) to
CONTINUE, so your dialplan continues after everything is complete (or, if
you finish everything within the routine, just let it end there).
Josh
On Wed, Apr 30, 2014 at 10:21 PM, Igor Dvorzhak idm...@gmail.com wrote:
Thanks, it almost what
requests to each server as needed, and I'm not even
sure if that is possible (and would require me to learn a LOT more about
opensips).
Thanks,
Josh
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New
With such a low amount of calls per month and with the extreme memory
limitations, it might be easier to write a script to pull out the data and
generate a static html page. Run it daily / weekly / whenever you need it.
On Thu, Apr 24, 2014 at 8:28 AM, binary dreamer
Instead of using CDR for this, could you get the info you need using
channel event logging (Asterisk CEL)? I have never used it myself - just
something I've run across in the past that seems like it might work for
this case:
https://wiki.asterisk.org/wiki/display/AST/CEL+Design+Goals
On Thu,
, but that's something else entirely...). There was mention of
checking against a DNC list, and ODBC would be good for this as well - just
put that into a table and match against it before making your outbound
call.
-Josh
On Wed, Apr 23, 2014 at 4:12 AM, James Sharp ja...@fivecats.org wrote:
On 4
want complicated, somewhere I have a very long GotoIf() that
includes an ODBC call and nested Math() functions...
-Josh
On Wed, Apr 23, 2014 at 11:17 AM, Doug Lytle supp...@drdos.info wrote:
I tried database access in the dialplan using the mysql() application
years ago, just to confirm I
How many seconds later does the file show up? Can you just throw in a
Wait() (maybe 1 or 2 seconds) and then do the Playback, or would even a
second or two of delay be an issue (or does it still not work)?
-Josh
On Wed, Apr 23, 2014 at 2:23 PM, CDR vene...@gmail.com wrote:
Dear friends
I
As a second possible solution, instead of Record, could you use
MixMonitor, then run StopMixMonitor and THEN do your Playback? That
should definitely make sure the recording file is closed and the file
handle released.
-Josh
On Wed, Apr 23, 2014 at 2:29 PM, Josh Metzger joshdmetz
into fancy legal issues about using an
autodialer when you accidentally call someone who doesn't want to be
called and they complain (dependant on jurisdiction).
-Josh
On Wed, Apr 23, 2014 at 2:36 PM, Steve Edwards asterisk@sedwards.comwrote:
On Wed, 23 Apr 2014, Steve Edwards wrote:
I tried
:
asterisk-users-boun...@lists.digium.com] On Behalf Of Josh Metzger
Sent: Wednesday, April 23, 2014 2:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help with a bug
As a second possible solution, instead of Record, could you use
MixMonitor
could
make it work, but for what I'm doing it really is probably the best option
(especially since it's on a pre-existing Asterisk install that was not
configured with ODBC support).
-Josh
On Mon, Apr 21, 2014 at 10:27 AM, Jonathan White j...@uvacity.com wrote:
I’m trying to use the asterisk
show application dial gives you all the
possible arguments for Dial, including some useful notes. Quite handy.
Josh
On Wed, Apr 16, 2014 at 9:22 AM, Kevin Larsen
kevin.lar...@pioneerballoon.com wrote:
Thank you guys – your advice was spot on. I will now reach out
earlier
Asterisk
and run dahdi_tool, is it showing you the circuits in an OK state?
Josh
On Wed, Apr 16, 2014 at 5:25 PM, st...@vanwambeck.net wrote:
Hi all,
I have a fresh install of Asterisk 11.8.1 and am putting a Digium TE435 4
T1 card in it for ISDN PRI. I can get the card to be recognised
to go
from zapata.conf to chan_dahdi.conf). It can be tedious, but it really
only took me a day or two (taking time to double-check my changes).
Overall, not a bad experience at all.
Josh
On Tue, Apr 15, 2014 at 4:37 AM, Lee, John (Sydney)
john@compuware.comwrote:
Hello,
I have been
, making such an effort a waste?
Thanks,
Josh
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header?
On Monday 22 July 2013, Josh Hopkins wrote:
Would it be possible to set the ringtone based on the number that was
dialed?
If the phones you are using allow the ringing tone to be changed by sending a
SIP header, yes.
Example of what the goal is:
Dial Denver number
Would it be possible to set the ringtone based on the number that was dialed?
Example of what the goal is:
Dial Denver number
Incoming calls ring with ringtone 1
Dial main number
Incoming calls ring with ringtone 2
We are currently using Digium D40, D50, D70 phones.
--
I am currently using AsteriskNow v2.
What I am trying to accomplish is having all calls from an area code go
directly to the person responsible for that area. While searching for a
solution for this I did come across a post that had a few examples. So Josh at
extension 1902 would receive
-- Executing [s@macro-auto-blkvm:4] ExecIf(SIP/1010-0161,
0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=Josh Hopkins)) in new stack
-- Feature Found: apprecord exten: apprecord
-- Executing [s@macro-one-touch-record:1] ExecIf(SIP/1010-0161,
0?Set(THISEXTEN=1010)) in new stack
: Wednesday, August 22, 2012 6:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] recording calls
you need to provide dial plan for macro-one-touch-record.
i think there is something which records outgoing only
On Wed, Aug 22, 2012 at 6:39 AM, Josh Hopkins
-- Executing [s@macro-auto-blkvm:4] ExecIf(SIP/1010-0161,
0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=Josh Hopkins)) in new stack
-- Feature Found: apprecord exten: apprecord
-- Executing [s@macro-one-touch-record:1] ExecIf(SIP/1010-0161,
0?Set(THISEXTEN=1010)) in new stack
I have been looking for the specs (format, bit rate, ect) on custom ringtones
for digium phones. Using the DPMA how would I deliver the ringtone to a digium
phone?
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On 8/20/2012 10:14 AM, Josh Hopkins wrote:
I have been looking for the specs (format, bit rate, ect) on custom
ringtones for digium phones. Using the DPMA how would I deliver the
ringtone to a digium phone?
https://wiki.asterisk.org/wiki/display/DIGIUM/DPMA+Configuration
All
ever tried this? Thanks,
/Josh
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Subject: Re: [asterisk-users] Voicemail Emails
Have you tried to insert the HTML code directly into the body?
Il 20/07/12 19:53, Josh Hopkins ha scritto:
Has anyone been able to make an html template for the voicemail emails. We
would love to customize them beyond just plain text. We have dome some
The most current patched Asterisk, along with the most current app_rpt,
can be found at
http://svn.ohnosec.org/svn/projects/allstar/astsrc-1.4.23-pre/trunk/
The code in the Digium SVN repository (at the link Steve provided) has
not been updated in three years.
On 3/9/2012 7:52 AM, Steve Totaro
viewpoint) to just set
up a second box with canonical 1.8 or 10 and trunk the two together.
Josh Freeman
On 02/23/2012 08:57 AM, Paul Belanger wrote:
Good morning,
There is a new patch up on reviewboard[1] right now for the removal of
app_rpt and chan_usbradio from Asterisk trunk. As it stands
http://www.asterisk.org/astdocs/node66.html
Thanks, never knew that!
Yes, I understand that it's not what you want, but that doesn't make
it a security concern. If Asterisk is publicly available on one
interface, making it available on another interface doesn't make you
less secure.
You
I don't get this. Didnt EVERYONE know it's insecure?
Can you read?
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As far as I know, Asterisk would use the default Linux/Unix routing
algorithms to send packets out, in which case yes: responses may not go
out on the same interface packets were received on.
E.g. if you receive packets with non-LAN IP addresses on eth0, while
your default route is set to
The primary goal was to upload audio for IVRs in the Asterisk GUI.
Thanks, if I don't use the GUI is it safe to exclude it from the build
(it is just that I want to avoid a bunch of other dependencies which
come with that module)?
--
It is indeed. This is already implemented in Asterisk I take it then? If
so, brilliant news!
More or less. I don't know if it's easy to trigger for specific
caller ID values, or for none. You might need to to a little
customization, but something mostly like what you describe is present.
I
All of that is true, but none of it appears to be a security concern,
specifically.
For you, may be, but from where I am sitting, I don't want to rely
solely on netfilter/iptables to protect me when I could physically
restrict Asterisk from binding to that interface (and answering such
In short - is this module essential for the running of Asterisk? What is
its function? Is there a help/list where I could find a description of
what it does? Thanks!
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Why do you see binding to 0.0.0.0 to be a security risk?
Purely because a response from Asterisk can be received as a result of a
connection on *any* interface on the system/machine. If I have Asterisk
confined to, say, 2 interfaces - eth0 (10.1.1.1) and eth1 (10.2.1.1)
then a request over a
While usually thread hijacking is not something that should be done,
in this case thank you for hijacking it as the OP on his original
topic was way off topic.
Why is that - I think I posted legitimate questions/queries with regards
to the installation, configuration and running of Asterisk
Your description sounds almost entirely like the existing call
screening, so I'm pretty sure you'll be able to accomplish it. Start
with call screening, and modify that to suit your needs.
It is indeed. This is already implemented in Asterisk I take it then? If
so, brilliant news!
I'd
Whats asterick?
I blame my spell checker! :-P
Do you have anything to offer in terms of help or advice on the
issues/questions I posted?
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I think you might want to split your questions first.
I thought that instead of creating a dozen different threads (and
clogging the ML in the process) it would be better to put everything
into one place - just pick the issue (or issues) you could address and
leave (i.e. delete) the rest
of the participant channel.
I suppose I could do this with a System call, something like
System(asterisk -rx confbridge record start ) - but is there a
better, less-roundabout way of getting there?
Thanks,
Josh
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I am trying to configure Asterick, having the following system setup on
the Asterick server:
* eth0 faces the external Internet interface, *but* it does not have IP
address (it has a private one given to it by my ISP's DHCP server);
* eth1 faces my internal network (say 10.1.1.0/24);
* tun0
If I understand correctly, turning off Call Screening in your Google
Voice configuration should directly connect incoming calls and eliminate
the need to press one.
JF
On 12/2/2011 11:59 PM, white hat wrote:
When a caller calls my google voice phone number, I must answer, wait
and press one to
that would get me most of the way there, but I'm constrained
to use an Asterisk 1.4 system which doesn't appear to have that application.
Anyone have any ideas on how I might make something like this work?
Regards,
Josh
On 10/10/2011 2:59 PM, Cassius Smith wrote:
On 10/10/11 10:40 AM, Josh Freeman cpe.jfree...@gmail.com wrote:
Hello,
I'm looking at a scenario in which, to make it work, I'd need to dial
into a remote conference from within a local MeetMe room. That might
include being able to dial
not understanding correctly?
I know there was a bug where all calls were recorded as 'NO ANSWER', I
was plagued with that until updating to 1.6.2.6, this does not seem to
be related.
Any feedback is appreciated.
Thanks,
Josh McAllister
send an e-mail to our cell phones so we knew about it right away. It seems
like there would be a market for this type of tool, but I can't seem to find
any service or software that will do what I'm wanting. Any suggestions?
--
Josh Hunholz
Has anyone ever gotten an Adit 600 to work with Asterisk1.4 via MGCP.
Asterisk keeps giving me the following error in the LOGs:
[Jun 30 08:32:59] NOTICE[26785]: chan_mgcp.c:1726
find_subchannel_and_lock: Gateway 10.0.0.245' (and thus its endpoint
'*') does not exist
MGCP Config:
not well
versed in *nix) and get out of rpm hell.
b) install headers:
apt-get install build-essential linux-headers-`uname -r`
c) still having problems? Install all the dev packages for zaptel:
apt-get build-dep zaptel
Thanks,
Josh Fuller josh.ful...@telus.com
The views expressed
links to have conversations between
Florida and Ontario
almost fifteen years ago. It's more like a two-way radio than a telephone but
it works very
well and is win/lin cross-platform.
[1] http://speak-freely.sourceforge.net/
[2] http://speex.org/
Thanks,
Josh Fuller josh.ful
session-timers on
certain call
flows in a modular dial plan.
Thanks,
Josh Fuller josh.ful...@telus.com
The views expressed in this e-mail are mine alone and do not necessarily
reflect the
views of my employer.
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Thanks,
Josh Fuller josh.ful...@telus.com
The views expressed in this e-mail are mine alone and do not necessarily
reflect the
views of my employer.
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much you want to fix it up.
Thanks Everyone,
Josh
---
When I insmod/modprobe tor2.c, however, I get a segmentation fault, and I can't
use the driver, or even unload it. The only way to remove the driver is to
reboot the machine.
develop
For such a simple application I'd use AstDB to avoid having to hassle with
an external database (and also means this sort of dialplan will work even on
embedded/slimmed Asterisk boxes that may not have db modules
loaded/available). In any case, what Tilghman said is what I'd suggest as
well.
channels vs. IAX channels?
--
Josh Richards - Grover Beach, California US
[EMAIL PROTECTED] (don't forget the middle 't' initial when writing)
http://blog.joshrichards.org/
805/471-6923 (cell)
Geek Research (Technology Management Consulting) -
http://www.geekresearch.com/
Support These Nifty
work with Linux correctly in regards to IRQs)
Optional:
On board hardware RAID
miniPCI
CF slot
IPMI interface
--
Josh Richards - Grover Beach, California US
[EMAIL PROTECTED] (don't forget the middle 't' initial when writing)
http://blog.joshrichards.org/
805/471-6923 (cell)
Geek
(not all) people phone me (isdn-incoming) DTMF is not
recognized.
How come?
Either it works for a particular configuration, or it doesn't.
It doesn't make sense to me that it works sometimes...
--
Josh Richards - Grover Beach, California US
[EMAIL PROTECTED] (don't forget the middle 't' initial
Hello,
I would like to put 1 asterisk box in Country A and 1 asterisk box in Country B.
Let's assume :
- Asterisk box in country A = GWA
- Asterisk box in country B = GWB
- Calling party number (located in country A) = CgPNA
- Called party number (located in country B) = CdPNB
- Second Called
, but this was easy and it worked for me. What's next
on my to-do list is trying to cover up the TellMe jingle before it
starts the VoiceXML app. If anyone would like to help clean up the
code, or has a better way of interacting with the Asterisk manager,
please let me know.
Thanks,
-Josh
We have many 7940s and 7960s in use with standard 802.3af gear using
custom cables that reverse on two pairs. There are instructions on how
to make them here:
http://www.voip-info.org/wiki/index.php?page=Cisco+POE
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
@ hermes on a i686 running Linux
on 2006-07-11 19:28:30 UTC
There's plenty of disk space on the partition and permissions are OK.
I'm a little stumped.
Any ideas?
Thanks,
-Josh C.
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fine with the account codes.
I'm stumped. Anyone have any ideas or pointers?
Thanks,
Josh
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there and would be
willing to help in it's formation.
Josh
Message: 15
Date: Wed, 5 Jul 2006 14:00:35 -0600
From: Douglas Garstang [EMAIL PROTECTED]
Subject: [asterisk-users] Asterisk in Seattle
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID
there.
Josh McAllister
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Danish Samad
Sent: Thursday, June 08, 2006 8:25
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] how to
identify agi crash cause
Hi,
I have a custom agi which at times does not exit
to Asterisks screen make sure you detatch the screen, and do not Ctrl-C or
exit the asterisk console as that will shutdown asterisk.
Josh McAllister
From: Danish Samad
[mailto:[EMAIL PROTECTED]
Sent: Thursday, June 08, 2006
11:37 AM
Hi,
Thanks for your reply. Dont the messages logged
Title: Meetme and authentication
Perhaps youve already figured this
out, but I posted an example dialplan and small Perl AGI that would resolve
this for you. As it happens this was posted the Friday before you sent this.
Look for a posting from me on Friday, May 12th.
Josh McAllister
${array[0]}=${array[1]}
done
This will export various ENV variables containing the info asterisk is sending.
For more info look at the BASH resources on the Asterisk AGI page of
voip-info.org. (http://www.voip-info.org/wiki-Asterisk+AGI)
Josh McAllister
-number-accepted,'');
return 'User';
} else {
$AGI-stream_file(conf-invalidpin,'');
}
}
return undef;
}
What can I say, I was bored.
Enjoy,
Josh McAllister
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf
loaded up with perl
AGI scripts and never skipped a beat. FWIW, these servers have 4G ram, and run
64bit RHES. Either way, glad I could get you closer to the end.
Josh McAllister
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Damon
of understanding can lead to nasty things like
replicating the wrong way.
Note, that this can be used as a very simple means of providing warm
standby * servers as well. Coupled with something like mon, you can
provide for automatic failover as well.
Josh McAllister
-Original Message-
From: [EMAIL
Hi Josh -
Another approach you may want to consider for data redundancy that
does not rely on MySQL's finicky replication stuff is DRBD. Think of
it
as RAID-1 across Ethernet. I have used it in production on some VERY
busy ( 1200 qps) MySQL servers for a couple years with no problems
, then decoding and allowing the archives to be viewed
online along with all relevent call details.
Hmm... Interesting.
Josh McAllister
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin
Anderson
Sent: Thursday, May 04, 2006 1:12 PM
To: 'Asterisk Users
Just a shot in the dark... but have you tried Answer() before
Playback()?
Josh McAllister
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom
Engleward
Sent: Monday, May 01, 2006 11:43 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users
cID Name in the CDR records all along as well.
Eric -- Go ahead and give it a shot... even if you are getting the cID number.
This will likely fix your problem.
Josh McAllister
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Asterisk
is not multi-threaded.
For something like this, I think you'll find 1 instance of a single
script much easier to track and debug than a whole bunch of instance of
an AGI script.
Josh McAllister
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon Farmer
Sent
Just saw this on Cisco's software download site:
7941/61 IP Phone SIP phone load - for CCM v5.0
Has anyone used this with Asterisk yet?
Josh
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make: *** [subdirs] Error 1
Thanks,
Josh McAllister
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I'm sure there is more than 1 way to do this, but the first thing that
comes to my mind is to set a channel variable with the exten # at the
top of your extensions macro. Then use that channel var instead of CLID.
Josh McAllister
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
is correct, it would be of great benefit to know exactly which
events are in which category.
Josh
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu
Sent: Tuesday, April 04, 2006 1:35
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk
.
Thanks,
Josh McAllister
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,Goto(menu,1)
exten = s,2,Hangup
;
exten = menu,1,SetVar(count=0)
exten = menu,2,Answer
exten = menu,3,Background(silence/1)
exten = menu,4,Background(josh/welcome-msg)
exten = menu,5,Background(silence/5)
exten = menu,6,SetVar(count=$[${count} + 1])
exten = menu,7,GotoIf($[${count} 1]?4) ; Repeat 3
On Mar 2, 2006, at 2:15 AM, Vahan Yerkanian wrote:
Reboot once again and it picks up the new config. Two-step
provisioning takes a couple of reboots to insure the device has
reconfigured itself. Applies to 2100, 3000, 841 and 941 models.
I've had good results on our 942 by setting the
(now that I've remembered which address is subscribed to this list)
Does anyone with one of these phones have any sort of presence
working? I'm looking to monitor the DND state of the phones, if
nothing else. Setting the SIP-B bit enables SUBSCRIBE/NOTIFY, but
the dialog package is the
I've seen a similar problem before. Span 3 was throwing errors for
(what seemed to be) no reason at all. After some testing it seemed
that the number of errors thrown on Span 3 had a relationship to the
temperature inside the servers. After installing additional cooling
the errors had
.
- Josh Harington [EMAIL PROTECTED]
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Hi all,
I'm kinda new with asterisk stuff.
I'm running a Debian with asterisk and a digium X101P clone card in country #1.
Since I'm going to work in another country (country #2), I would like
to setup another Asterisk server + 1 FXO device in #2 as well as in
#1.
However I'm looking for a small
On Jun 6, 2005, at 4:39 PM, [EMAIL PROTECTED] wrote:
I want to manage this dialplan variable for each extension separately,
unfortunately this doesn't work:
**77,hint,DS/splat${CALLERIDNUM}
Do you have an idea for that?
Is there an easy place to patch it in asterisk 1.0.7 stable?
Will it be
On Jun 4, 2005, at 4:52 AM, [EMAIL PROTECTED] wrote:
I would like the SNOM extension light to permanently
reflect the current toggle status of my application logic/asterisk DB
variable.
There's a phantom device in bristuff that can be used for this sort
of thing. When you toggle the
Has anyone sucessfully set up wIP Phone for asterisk?
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On Apr 17, 2005, at 12:23 AM, Lance Grover wrote:
I have rebooted the phone and restarted asterisk after each change.
Did you do it in that order? If so, that is probably a source of
trouble (you should restart or reload asterisk before the phone boots,
not after).
--
Joshua P. Dady
(boy mail in this list piles up fast when I can't check it)
On Apr 8, 2005, at 10:03 AM, Michael George wrote:
- It appears that the extension used with the hint must be the same
as the
extension used to dial that channel. So if extension 22 will ring
Zap/2,
then exten = 22,hint,Zap/2 will
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