Of Joshua Colp
Sent: Tuesday, February 17, 2015 4:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Callfile problem - Unable to find codec
translation path from (nothing)
Justin Killen wrote:
snip
Whenever I try to copy this callfile into
/var
Hi,
I copied a setup from an older 1.8.5 installation to an 11.15 installation, and
I'm having problems getting call files to work. Here is the extension setup
I'm using:
[outbound-swift]
exten = _[a-zA-Z].,1,Answer
exten = _[a-zA-Z].,n,Playback(AAA/check_ip_failure)
;exten =
I have a dialplan (freepbx) that plays a busy signal in-band when an extension
is busy (before an Answer). Stripped down, it looks like this:
exten = 1005,n,PlayTones(busy)
exten = 1005,n,Busy(20)
Note that there is no Answer() prior to this. Our trunk is a PRI.
When I call into this extension
I have two servers, each connected to the PTSN via PRI. When I call from site
A (951-999-) to site B (555-1212) and the phone at site B is on the phone,
I hear the normal ring tone for about 20 seconds, then the message all
circuits are busy now. please try your call again latter followed
If you use Playtones you should put an Answer and a Wait(1) before the Playtones
I recommend using the Hangup app instead. Busy would be Hangup(17).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Wednesday, July 09
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Wednesday, July 09, 2014 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI congestion instead of busy
I tried changing the dialplan to use Hangup(17
Is there a channel variable / status indicator / function that indicates if the
current channel has been answer()'ed?
-Justin
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Okay, I think I need a sanity check here - If I call a person that's on the
phone, I should get a busy signal.
Now more specifically, a call comes into the pbx via PRI. The destination
dialplan runs busy(20). Now, the PRI causecode should get set to 17 (user
busy) so that the originating end
to the caller you would use Hangup(17)
and let the caller's switch generate the busy tone.
If the dialplan has already answered the call, then you might want to use Busy
or Playtones.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin
Plus, some traffic got split off into the app-dev list (and there's the dev
list).
-Justin
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Ron Wheeler
Sent: Sunday, March 02, 2014 6:10 PM
To:
Using Dahdi/PRI, I end up with channel names like 'DAHDI/i8/9995551212-4d6B',
but when I do a 'core show channels' it cuts off those names to only
'DAHDI/i8/9995551212-'. This is the same for AMI.
Is there a way to get the full channel name within AMI?
I'm using asterisk 11.7.0
Thanks,
After posting this, I ran across 'core channel show concise', which gives the
data in a more machine friendly format.
-Justin
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Thursday
Another option is to use an MRCP server like UniMRCP along with the Cepstral
plugin. One very nice thing about this approach is that there is less
'cepstral version' - 'asterisk version' dependency, which is a problem with
the current app_swift module (each app_swift version is designed to
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Thursday, November 14, 2013 3:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] recieve fax from PRI using spandsp 65% failure rate
Hi,
On one of our servers, we're
Hi,
On one of our servers, we're having problems with incoming faxes. The
connections come in through a PRI into a Digium TE820 card. 'fax show stats'
yields the following:
FAX Statistics:
---
Current Sessions : 0
Reserved Sessions: 0
Transmit Attempts: 0
Receive
] analog phone digit delay
I imagine setting up a catch-all extension pattern is your best option. That
is what most seem people do.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Wednesday
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Thursday, July 11, 2013 12:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
So my only two options then are:
1) Have the timeout be so
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Thursday, July 11, 2013 1:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
No, I understand - maybe I'm not explaining myself well.
Yes, I can
-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Thursday, July 11, 2013 3:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
Right, but when
] On Behalf Of Justin Killen
Sent: Thursday, July 11, 2013 4:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
They won't catch, no (because of priority), but they do match, which is enough
to trigger the 3 second timeout
] On Behalf Of Chad Wallace
Sent: Thursday, July 11, 2013 2:05 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] analog phone digit delay
On Thu, 11 Jul 2013 13:53:27 -0700
Justin Killen jkil...@allamericanasphalt.com wrote:
They won't catch, no (because of priority), but they do match
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday, July 08, 2013 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
On Mon, 8 Jul 2013, Justin Killen wrote:
I have an installation
Okay, after enabling DTMF logging, what I see is a handset being picked up, 7
digits being pressed in 4 seconds, and then 3 seconds input is determined to be
done and the call is processed (to the catch-all 'bad-number').
What I don't understand is that if the digit timeout is set to 5, then
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
On Mon, Jul 8, 2013 at 12:14 PM, Justin Killen
jkil...@allamericanasphalt.commailto:jkil...@allamericanasphalt.com wrote:
I have an installation that has analog phones connected via T1 channel
, if there is an ambiguous match
*/
static int matchdigittimeout = 3000;
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Wednesday, July 10, 2013 3:55 PM
To: Asterisk Users Mailing List
-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Wednesday, July 10, 2013 4:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
It seems likely that this is exactly what
I have an installation that has analog phones connected via T1 channel banks.
I'm getting complaints from users that they will enter a partial number (eg
91213), then turn away to get the next few digits, and the system will start
dialing before they have a chance to put in the rest of the
The channel banks are Adtran TA-624's using ESF/B8ZS. When a handset is picked
up, I can see the offhook in the asterisk console, so it looks that the channel
is immediately connected through the channel bank (not delayed until after
digits are dialed), so it looks that overlap dialing isn't a
http://192.168.2.10:8080/test has an IVR service.
could you please help me with an guide to do the configurations ?
Many Thanks
Luke
From: Justin Killen jkil...@allamericanasphalt.com
To: luke devon luke_de...@yahoo.com; Asterisk Users Mailing List -
Non-Commercial
1) I'm sure it's possible, it's just a matter of talking to the mobile network
in question and finding out what kind of options they have available.
2/3) Once asterisk gets the call, it can redirect it to just about any endpoint
you'd like it to. For more information on asterisk's IVR
If you just want the url to be opened (perhaps to update a counter via a web
service or cgi script), you can do this:
system(wget http://;)
or
system(fetch http://...;)
-Justin
From: asterisk-users-boun...@lists.digium.com
You'd probably be better off sending this to the dev list (asterisk-dev)
Justin Killen
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Optical Phoenix
Sent: Tuesday, March 05, 2013 5:56 PM
To: asterisk-users@lists.digium.com
Subject
/system.conf and /etc/asterisk/zaptel-channels.conf
-Justin Killen
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine
Elharit
Sent: Wednesday, February 20, 2013 10:33 AM
To: Asterisk Users Mailing
between them,
adding the routes for the private networks to cross thru the tunnels.
Justin Killen
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank
Sent: Thursday, February 07, 2013 9:49 AM
To: Asterisk
Or if it's just a couple phones, you might be able to setup a vpn connection
directly on the phone itself - have it vpn into 'HQ' and get an address on that
network. I'm not sure which phones you're using though or what phones support
that setup.
Justin Killen
-Original Message-
From
You are correct, this is not an asterisk question. What I would suggest would
be to run your script outside of asterisk and debug the connection. Looking at
the php doc page for fsockopen
(http://php.net/manual/en/function.fsockopen.php), I see this example:
?php
$fp =
a
matter of calling the web service using curl,and then wait for the response?
what am I missing?
Christian Savinovich
VoIP Telephony Consultant
646-982-3572
Original Message
Subject: Re: [asterisk-users] problem to socket programming in AGI
From: Justin Killen
jkil
Valer,
Thank you for the advice - I have support tickets open with Adtran and Digium
and we are tracking down the issue. Hopefully it doesn't come down to adding
more hardware, but I'll keep that in mind.
-Justin Killen
From: asterisk-users-boun
I have several adtran 624 with 24 FXS ports hooked up to analog phones. The
adtran is connected to asterisk via a channelized T1 into a digium TE820. I
have hardware echo canceling enabled on all channels/spans, but there is still
echo on the lines for both calls out of the trunk, as well as
echocancel=yes
context=from-internal
callprogress=no
callgroup=
callerid=John Doe 3884
busydetect=no
busycount=7
accountcode=
channel=73
...
-Justin Killen
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This is highly confusing. It would be nice if at least the display gave the
configured value as well.
-Justin Killen
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russ Meyerriecks
Sent: Thursday, December
me through a
series of 'I call you, you call me' scenarios and then the tool would tell me
what settings to use?
Justin Killen
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A bit off topic, but does anyone know what happened to the freepbx.org website
and when it's coming back online?
-Justin
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New to Asterisk? Join us for a
Is there a way to specify default files to use for new mailbox creations? For
example, when a mailbox's directory structure is created, there is no greeting,
unavailable, or busy messages, so the incoming calls get the message: The
person at extension XX is not available. I'd like to be
Those are asterisk downloads, not dahdi downloads
Justin Killen
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Oyarzo
Sent: Wednesday, November 14, 2012 4:28 PM
To: Asterisk Users Mailing List
In http://packages.digium.com/centos/ there is not yet a centos 6 branch (Nor
is there a RHEL 6 branch). Centos 6.0 was release in July of 2011 - is this
something that Digium is planning on supporting? Or is there a different URL
that I'm not aware of for firmware packages?
-Justin Killen
filtering system, so sometimes things get blocked)
But still when I run dahdi_hardware, I still only see the TDM800P card
-Justin
-Original Message-
From: Shaun Ruffell [mailto:sruff...@digium.com]
Sent: Thursday, November 08, 2012 7:24 AM
To: Justin Killen
Subject: Re: [asterisk-users
I just installed a TE820 octal span T1 card, and it's not showing up in
dahdi_hardware output. This was installed into a test machine that already has
a TDM800P card in it, and that one is showing up and working fine. Is there
some kernel module that I'm missing?
Lspci:
05:04.0 Ethernet
I'm looking for an fxs - sip gateway/router/switch for about 100 existing
analog phones. I'd like to get this done cheaply, but I want to make sure that
whatever we buy works well with asterisk as well. As far as I can tell, digium
make no such device. The only ones I've been able to find
What would be the advantage of using 100 single units vs. just buying VoIP
phones? That doesn't seem very cost effective to me in the long run.
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New to
] high capacity analog - sip gateway
On Thu, Oct 25, 2012 at 1:21 PM, Justin Killen
jkil...@allamericanasphalt.commailto:jkil...@allamericanasphalt.com wrote:
I'm looking for an fxs - sip gateway/router/switch for about 100 existing
analog phones. I'd like to get this done cheaply, but I want
I think if we were to go to VoIP phones, one thing that we would have to
consider very highly in a phone would be that they have VLAN settings and a
built-in Ethernet hub/switch so that we can just inject it into the user's
computer LAN connection. The cost and time of rewiring some of these
Swift() is an asterisk wrapper around the text-to-speech engine cepstral. Looks
like this is a dev issue - I'll start a new thread on the dev mailing list.
Justin Killen
Senior Programmer / Analyst
All American Asphalt
951-736-7600 x 2060
jkil...@allamericanasphalt.commailto:jkil
is having a problem, correct?
Justin Killen
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Tuesday, May 22, 2012 8:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I have and automated call-in dispatch system where hundreds of people call in
daily for 2-3 minutes each. The extension is set up to get their information,
then text-to-speech the dispatch information (via odbc). It then loops 5 times
then ends the call. These calls are being handled by an 8
Hi, all.
I am attempting to get res_odbc working with Informix (server version 11.10.UC1
on an AIX 5.3 box, Informix csdk (client) version: 3.70.UC3DE on a CentOS 6.0
box). I am able to successfully connect and run a query via isql, but if I
attempt that same query via the dialplan, I get
...@lists.digium.com] On Behalf Of Justin Killen
Sent: Monday, August 22, 2011 10:17 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] res_odbc with informix
Hi, all.
I am attempting to get res_odbc working with Informix (server version 11.10.UC1
on an AIX 5.3 box, Informix csdk
Hi All,
We just bought a sip based PA setup here with the intention of hooking
it into our existing asterisk (1.4) setup. It works as expected when I
dial it's extension, but I want to have system generated speech played
based on some action (using cepstral, which is already installed and
.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin
Killen
Sent: Monday, April 20, 2009 2:14 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk PA system with cepstral
Hi All,
We just bought a sip
we tried to Goto(ids, 4555, 1) would we get directed to
extension i in the extensions context or would the call be dropped
completely?
Moj
Justin Killen wrote:
I have an application where a call-in user is prompted to enter an
identification number for schedule information. That id
I have an application where a call-in user is prompted to enter an
identification number for schedule information. That id number is setup
as an extension, and if that extension doesn't exist, it tells them that
they are not scheduled, then loops back to ask for the id number again.
My problem is
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