:39 system1 local3 FXO: On Hook
Dec 1 17:23:39 system1 local2 AUD: Stop PSTN Tone
Dec 1 17:23:39 system1 local2 FXO: Stop CNDD
Regards,
Justin
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Check out
Trying to configure Asterisk 11 Cert with G722.1C. I have installed the
latest binary for Siren14:
srv-echo*CLI> siren14 show version
Digium Siren14 Module Version 11.0_1.0.5 (optimized for opteron_sse3_64)
According to this list post in 2012 Asterisk supports G.722.1 Annex C (also
known as Siren
api-digital.com --
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n, one of the origin legs? Is there a way detect
this condition in the target context ([outbound-swift]), or better yet, verify
the other leg is attached before starting the logic?
-Justin
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-bou
the translation into doing the
right thing?
-Justin
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I would love to run Asterisk on a BSD system. I do not know of any developers
actively working on Asterisk on a BSD platform, though my knowledge isn't
comprehensive.
It may be worth talking to the people doing the packaging for various BSD
platforms, to see how involved they are, or if they
busy
tone).
Is there anybody out there who has experience with reading/analyzing IDSN trap
logs (Q931) that can help me narrow down where the issue is and how to fix it?
Thanks,
-Justin
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The description of busy() in the asterisk documentation wiki states:
"This application will indicate the busy condition to the calling channel."
Wouldn't 'indicate the busy condition' on a PRI channel imply setting cause 17?
-Justin
g end can play a busy tone, correct?
-Justin
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Is there a channel variable / status indicator / function that indicates if the
current channel has been answer()'ed?
-Justin
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New to Asterisk? Jo
dition has been
submitted as bug# 7706 - http://issues.freepbx.org/browse/FREEPBX-7706
The site A truck failover retry on hangupcause's that shouldn't be retried has
been submitted as bug# 7705 - http://issues.freepbx.org/browse/FREEPBX-7705
-Justin
/i8/951999-603'
in macro 'hangupcall'
[2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: == Spawn extension
(from-did-direct, h, 1) exited non-zero on 'DAHDI/i8/951999-603'
[2014-07-09 10:47:03] VERBOSE[7133][C-0c69] chan_dahdi.c: -- Hungup
'
of 17 (user busy). I suspect this is causing
site A to get the "all circuits are busy now" message instead of a busy signal.
I thought calling Busy() would cause the PRI cause to get set when used on a
c
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,
To follow up the discussion - yeah, it's not RAM, or at least not directly.
I'm so used to looking in the asterisk logs I didn't think to look at
/var/log/messages:
Feb 10 09:10:45 telephone-retsof kernel: [35734.705648] asterisk[11215]:
segfault at ffa2048e ip b70a3def sp b540a000 error 4 in
On 14-02-10 9:46 AM, Mike wrote:
> What log entries are leading you to think that you're running out of RAM?
None. It's just my guess. The log doesn't show anything except Asterisk
restarting.
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RAM and have it help, because it's 32-bit. I intend to move to a
64-bit machine, but I was hoping to wait until summer. Does anyone have any
immediate tips for dealing with this sort of rush?
Justin Sherrill - American Rock Salt
P: 585
.conf to
Dial() the sip endpoint whenever the extension is dialed.
Justin Hester
Digium, Inc. · Technical Trainer
445 Jan Davis Drive NW · Huntsville, AL 35806 · USA
ph: +1 256 428 6238
Check us out at: http://digium.com · http://asterisk.org
On Mon, Feb 3, 2014 at 5:45 AM, Raghav Goud wrote:
hat though, what does the CLI tell you if you do a
NoOp() after having Set() the Caller ID function [1]?
[1] Something like;
exten => _9NXX,1,Set(CALLERID(name)=mycompanyinc)
same => n,NoOp(The caller ID has been set to ${CALLERID(name)})
same => n,Dial(SIP/att/${EXTEN:1},80)
Hope
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
7;m using asterisk 11.7.0
Thanks,
-Justin
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uniMRCP can be found here (although it
seems to be having issues ATM):
http://code.google.com/p/unimrcp/wiki/asteriskUniMRCP
-Justin
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jg
Sent: Friday, J
Turns out I had my timing sources setup incorrectly, so the te820 wasn't
synchronizing with the Telco. Turns out the spandsp FAQs are pretty good for
diagnosing: http://www.soft-switch.org/spandsp_faq/ar01s08.html
-Justin
From: asterisk-users
ebugging (fax set debug on), but that doesn't seem
to do anything either (perhaps I need to raise asterisk's debug level as well?).
What steps should I try next?
Thanks in advance,
-Justin
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ntaining
it would be high. And of course, I'm using FreePBX (I guess I could put in a
request to change this for FreePBX, but then I'm sure there's others like
AsteriskNOW that would also need to change, so it makes more sense having this
within asterisk itself).
-Justin
the result
is that 3 seconds later they will flow into error handling, where I think the
intension is that they should get a full 8 seconds.
-Justin
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
I am describing is compounded by
the fact that the patter is _X. instead of _X but the core issue is the same -
only getting 3 second inter-digit timeouts instead of 8.
-Justin
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On B
8 seconds), but timeouts are
valid extensions are excessive (8 seconds), and we get a fancy message.
It's a shame that reasonable timeouts and a nice message are mutually exclusive.
-Justin
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@
e and things are
running smoother.
Thanks Eric for your help on this - you helped me to track down the cause of
the issue and provided a work-around, which is much appreciated.
-Justin
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists
e mutually exclusive.
--Justin
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Thursday, July 11, 2013 7:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [aste
Okay, so I is no good. Does anybody else have a work-around for this?
-Justin
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, July 10, 2013 1:43 PM
To: Asterisk Users Mailing
It seems likely that this is exactly what is happening. I'd rather not change
the code though, but rather fix the dialplan. I'm thinking using the 'i'
extension would work just the same - would there be a reason to use a wildcard
pattern match instead of i?
-Justin
-
isk will wait for 8 seconds instead
of 3? The next question then is how to accomplish this without using the
wildcard (and how to change it in freepbx).
-Justin
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On B
n extension
(from-internal, 1909996, 7) exited non-zero on 'DAHDI/96-1'
[2013-07-10 09:22:46] VERBOSE[12753][C-0002ec16] pbx.c: -- Executing
[h@from-internal:1] Hangup("DAHDI/96-1", "") in new stack
[2013-07-10 09:22:46] VERBOSE[12753][C-0002ec16] pbx.c: == Sp
Values for the timeouts just before the 'cannot complete as dialed, please try
your call again':
absolute: 0
digit: 5.000
response: 10.000
I've enabled DTMF logging to try to get a better log for interpretation.
-Justin
-Original Message-
From: aste
7; which is obviously not what I want. I would expect to
see something like 'dialtimeout' or 'interdigittimeout'.
-Justin
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jg
Sent: Monday, July
f the dialing string. Is
there a way to increase this delay? The only use these 4 dialing patterns:
Internal 3 digit numbers
91 XXX XXX (for backwards compatibility)
9 XXX (also for compatibility)
XXX
I'm using the freepbx distro if that helps. Asterisk 11.2.
Thanks
I'm unfamiliar with Voice XML. Just looking around a bit, it looks like Voxy
does exactly what you want - I'd look into that:
http://sourceforge.net/projects/voxy/
From: luke devon [mailto:luke_de...@yahoo.com]
Sent: Thursday, June 06, 2013 1:37 AM
nstall complete Asterisk'? I guess this depends on your
definition of 'complete'. My advice to you as a start/test would be to install
asterisk and all the sample configurations and then start playing around with
connections.
-Justin
From: asteris
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
[mailto
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
Su
aptel/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 Ma
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
-O
a vpn 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: Ast
Yes, I think curl would probably be a better option than trying to use sockets
directly, but if the socket won't connect it doesn't really matter what higher
level method is used.
-Justin
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-
and see what the
error string coming back is.
-Justin
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Muhammad
Sent: Monday, February 04, 2013 5:07 AM
To: Asterisk Users Mailing List - Non-Commerci
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-
on with asterisk, or if it's something I should first take
up with adtran?
I'm using dahdi 2.6.1, asterisk 10.10.0.
Thanks in advance,
-Justin
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8* for that one) keeps
going in a loop - downloads updater, saves it, formats the filesystem,
downloads the new bootROM, and then repeats. There's no error on screen and no
successful upload of logs to show an error.
Has anyone updated these models before and seen this?
Justin Sherrill - Ameri
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
=800
echocancelwhenbridged=yes
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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un the tool and then it could guide 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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"No Answer" is working fine when enabled. I
was looking at the sip.cfg but don't know exactly what to look for, can you
give me a hint to where would i find that option?
Thanks,
On Wed, Dec 12, 2012 at 1:48 PM, Justin Sherrill
mailto:justin.sherr...@americanrocksalt.com>>
uot;forward no answer" working?
Justin Sherrill - American Rock Salt
P: 585-991-6825 F: 585-991-6925 C: 585-298-6826
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s
like an echo.
I think turning rxgain/txgain down may make a difference, but I haven't tried
it yet. Has anyone else experienced something similar?
Justin Sherrill - American Rock Salt
P: 585-991-6825 F
I'd like to be able to specify
default files to be used and copied into the voicemail structure, something
similar to /etc/skel for user accounts. Does anybody know if such a feature
exists and how to use
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 K
behind a web content 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:
[ OK ]
wctc4xxp:[ OK ]
xpp_usb: [ OK ]
This is on a FreePBX 1.817.210.58 install - I'm not sure what the dahdi t
h, is this advisable or will there be any issues using a
value of "1" (by default it is OFF) on the SPA3102.
Thanks,
Justin.
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New to Asterisk
problem/if they have this issue.
Name: Voicemail
Message Number: 5
Mailbox: 1
Caller ID: "S"
Caller Name: S XXX
Caller Number: X
Duration: 1:22
Date: 20121101_1116
The voice mail:
http://home.comcast.net/~jpiszcz/20121101/msg0004.W
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 lo
n as well.
-Justin
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Thursday, October 25, 2012 1:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [aster
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 Ast
Does anyone have any experience with either of these products/vendors, or any
suggestions for a different piece of hardware?
Thanks
-Justin
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New to Asteris
http://lists.digium.com/pipermail/asterisk-users/2012-February/270427.html
That worked for me with the polycom 3.x firmware; I haven't tried it with 4.0
firmware yet.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaverstyn, David
C
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.com<mai
astapp' variable is 'Swift', this would indicate that the cepstral
wrapper 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,
Okay, the next time it gets in this state I'll gather that information.
Justin Killen
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Monday, May 21, 2012 1:22 PM
To: Asterisk
same => n,Set(AAA_CHECKED_IN()="${comp_num}", "${get_param2}",
"${account_id}", "${AAA_OUTPUT}", "I", "${CALLERID(num)}", "${CALLERID(all)}",
"${UNIQUEID}")
same => n,Swift("${AAA_OUTPUT}")
sa
plan I immediately play back a ringing sound to the Strata which
>> causes it to then send through the real external callerid
>> information.
>> As such my original email was a bit off the mark, I get a blank
>> callerid name and the callerid number is the virtual internal
>>
llerid
>> information is properly acquired and recognized in libpri it would
>> simply be accessible in Asterisk in the 'CALLERID(all)' variable, but
>> it is always empty. Internal calls from an extension on the Strata to
>> an Asterisk extension show the cal
to
an Asterisk extension show the callerid as expected.
Does anyone have any tips on how to get Asterisk to use the callerid
passed through by the Strata?
Thanks!
Justin
chan_dahdi.conf (group 2 is used outgoing only):
[trunkgroups]
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
use
x27;t
invoke a change. Should I be using register instead?
This does work as expected with a similar setup in sip.conf.
Thanks,
Justin Korkiner
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N
I have used a Plantronics CS351N and a CS70N with Polycom IP550 desk units.
(both are single-ear units, in different forms) Each one needed a Plantronics
APP-5 to replace using a lifter.
They worked fine. The one complaint that I had from users is that the headset
beep to show that a call wa
try using the
extended BLF stuff (described here http://www.excaliburtech.net/archives/147
and here http://www.voip-info.org/wiki/view/Asterisk+presence)
gordu
On Thu, Dec 15, 2011 at 12:10 PM, Justin Sherrill
mailto:justin.sherr...@americanrocksalt.com>>
wrote:
This is one of those "Is anyone else doi
Out of curiosity, what is "the Polycom script"?
I obviously haven't moved from 3.2.x firmware yet.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Friday, December 16, 2011 4:45 PM
To: Aster
sh the light next to that
extension, but that's it.
Is anyone using a similar setup and seeing this? It's somewhat rare, but I
have an office location where everyone there likes to pick up other people's
calls, and they haven't been using a call queue like they oughta.
Justin Sh
I've noticed that if I have people on speakerphone at the two farthest
ends of our internal network, they will occasionally get a second or two
of feedback. (sounds like jingle bells) I'm figuring it's some very
slight amount of packet loss or jitter that isn't helped by the
speakerphone echo, ma
Asterisk will send the two SIP endpoints 'reinvite' messages, so that they talk
RTP directly with each other. Depending on your version of Asterisk, setting
the 'canreinvite' or 'directmedia' option may make a difference, since that
will keep the traffic flowing through the servers, and the pho
lists.digium.com
[mailto:asterisk-users-boun...@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 versio
tor=INFORMIX
[asterisk-connector]
Driver=INFORMIX
Description=INFORMIX connection to asterisk database on borg
Database=rmca_test
Server=borgnet
LogonID=
pwd=
CLIENT_LOCALE=en_us.8859-1
DB_LOCALE=en_us.8859-1
TRANSLATIONDLL=/opt/IBM/informix/lib/esql/i
I've had mystery reboots with Polycom IP550s - the culprit in both cases was
the network connection. Replacing the cat5 cable to the phone or changing the
attached port fixed it both times.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@
Queuemetrics is neat-looking. However, it requires MySQL, and I'm using
Postgres. Does anyone have a recommendation for a different product for
reporting usage that's not tied to MySQL?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lis
phones. I haven't figured out yet
if they need a different voltage, or even if they still work; they were not
responding when I replaced the attached phones.
Justin C. Sherrill - American Rock Salt
p: 585-991-6825 f: 585-991-6926
ve the realtime peer and
> create a peer in sip.conf this works !!
>
> On Tue, May 3, 2011 at 11:15 AM, Justin Case
> wrote:
>> On Mon, May 2, 2011 at 1:09 PM, Deepesh D wrote:
>>> Hello,
>>>
>>> I upgraded from asterisk 1.6.2.7 to asterisk 1.6.2.17.
>
On Mon, May 2, 2011 at 1:09 PM, Deepesh D wrote:
> Hello,
>
> I upgraded from asterisk 1.6.2.7 to asterisk 1.6.2.17.
>
> I have context=defcontext set in sip.conf. For each peer I have
> context=outcontext in the peer definition since I want outgoing calls
> from registered SIP peers to go through
On Tue, May 3, 2011 at 2:50 AM, Ernie Dunbar wrote:
> I'm kind of at a loss to diagnose problems like this, yet we get them a lot.
>
> - The ATA (Thomson 784 in this particular case) is logged into the
> Asterisk server. 'sip show peer' shows their IP address, port, and
> useragent.
> - The ATA is
>-Original Message-
>From: asterisk-users-boun...@lists.digium.com
>[mailto:asterisk-users-boun...@lists.digium.com] On >Behalf Of Carlos Chavez
>Sent: Saturday, January 15, 2011 2:02 AM
>To: Asterisk
>Subject: [asterisk-users] Asterisk stops responding
>
> I am having a problem with
> From: marvin horst [mailto:fivehor...@gmail.com]
> Sent: Tuesday, October 19, 2010 10:23 AM
> To: Justin Sherrill; asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] integrate Intertel Axxess with Asterisk
>
> How did the setup work as far as extensions on th
>From: asterisk-users-boun...@lists.digium.com
>[mailto:asterisk-users-boun...@lists.digium.com] On
>Behalf Of Danny Nicholas
>Sent: Wednesday, September 22, 2010 5:04 PM
>To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>Subject: Re: [asterisk-users] Asterisk- speech to text(Voice
umber a few times
when testing and received a voicemail prompt instead of their main message, a
few times.
Justin C. Sherrill - American Rock Salt
p: 585-991-6825 f: 585-991-6926 c: 585-298-6826
--
_
-- Bandwidth and Coloc
I have an Asterisk 1.6.0.28 system, with a queue called 'marketing'.
Everything appears normal, but the status of the members never changes from
'not in use', even if they are being rang or are in a call.
Members are added like so:
queue add member SIP/1406 to marketing penalty 0 as SIP/1406
What would the power have to do with the D Channel ? Isn't which channel
used a logical setting (as opposed to physical). I am not saying your wrong
I am just trying to understand why it happens.
On Mon, Jul 12, 2010 at 7:56 AM, C F wrote:
> I have found that sometimes shutting down the machine
Sure. If you write the dial plan correctly and your legacy PBX supports it.
On Mon, Jul 12, 2010 at 7:31 AM, Malvin Rito
wrote:
> Hi List,
>
>
>
> We’re planning to use Asterisk as our backend PBX for our legacy PBX
> where-in received calls from legacy PBX can be transferred to Asterisk PBX
> e
It's not as easy as getting a switch and buying and selling minutes. You
have to learn what issues there are out there, day to day problems etc. You
will want to learn the protocols that you will be using (SIP/H.323 etc.). I
have two "carriers" that know less about SIP than I do. Before you jump in
On Wed, May 5, 2010 at 7:10 PM, Adrian Marsh wrote:
> Anyone have any experience with a Japanese local VoIP termination
> supplier?
>
>
>
> I’ve emailed a few companies looking to setup some PSTN to SIP and SIP to
> PSTN termination, but no luck so far.
>
>
>
> Thanks,
>
>
>
> Adrian
>
>
>
>
Try
Have you tried the SNOM forum ? They would probably have more info for
you http://forum.snom.com/
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 13 April 2010 10:12
To: asterisk-users@lists.digium.com
Subject
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