Darren
On 28 August 2016 at 10:20, Hooman Fazaeli <hoomanfaza...@gmail.com> wrote:
>
> Hi
>
> To troubleshoot FreeBSD panics triggered by ISDN load on an asterisk
> system,
> we are looking to buy an ISDN call generator/simulator device.
>
> The minimum requirements in
On Mar 22, 2013, at 5:22 AM, Florian Wolters flor...@florian-wolters.de wrote:
So I did setup another Extension leading me to a MeetMe conference to at
least listen to some MoH while waiting for the 15 Minutes to exceed. This
showed the same behaviour. After exactly 15 Minutes, the call is
both would be appreciated.
if you can send me a backtrace, that'd be great
On Jun 22, 2012, at 8:06 PM, Jeremy Kister wrote:
On 6/20/2012 8:24 AM, Darren Sessions wrote:
I just finished replying to your direct email (which you can disregard
now as this seems to be a different problem). I'm
Hi Jakob,
I just finished replying to your direct email (which you can disregard now as
this seems to be a different problem). I'm pretty sure I know what the issue
is, but I'll have to get back to you later this evening (my time).
- D
On Jun 20, 2012, at 4:41 AM, Jakob-Matthias Böttger
Hi folks,
Just a note to let everyone know I've finally finished up the new BETA release
of app_swift (now v3.0.1 b1).
This release introduces some pretty major changes to app_swift such as:
- The entire code-base has now been unified and the build system auto detects
which Asterisk version
T.38 is tolerant of most network conditions, ... the challenges in getting
reliable performance are usually limited to getting the interop right once, but
the absolute success rate will depend on the quality of your T.38/PSTN
gateway's fax implementation. In general terms, T.38 is actually the
On Feb 15, 2012, at 4:03 PM, Olivier wrote:
2012/2/15, Darren Nickerson darren.nicker...@ifax.com:
T.38 is tolerant of most network conditions, ... the challenges in getting
reliable performance are usually limited to getting the interop right once,
but the absolute success rate will depend
eventually.
-Darren
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Seasons Greetings!
- Darren
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We've been happy with the polycom IP 7000.
Darren Wiebe
On Nov 30, 2011 1:40 AM, virendra bhati virbh...@gmail.com wrote:
Hi Faisal,
Thanks for reply but I want hardware wase VoIP device. If know please
gussed me. From google I fould the list of below devices but I am not sure
time.
Enjoy,
- Darren
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asterisk-users
=Main_Page if you want to do
something similar. Why try to make Asterisk into something it's not
intended to be? Just use your firewall for what it's good at.
--
Darren Wiebe
On 7/23/11 11:38 AM, CDR wrote:
I beg to differ. Digium is hiding from the real world and somebody is
going take the software
interested the information is available here:
http://fail2ban.aleph-com.net/fail2ban_sharing If you're interested in
the server code just drop me an email.
Darren Wiebe
dar...@aleph-com.net
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You could use a sip proxy front end like Kamailio.
Sent from my iPhone
On Nov 18, 2010, at 7:39 AM, Antônio Theóphilo anto...@freeddom.com wrote:
Hi All
Does anyone know about any tool that does to Asterisk what mod_jk does for
JBoss/Tomcat: a load-balance/failover server that is
Well, the downside to wav files is the disk i/o. Asterisk will and does
translate the audio frames from ulaw to whatever other codec.
Sent from my iPhone
On Oct 24, 2010, at 9:42 AM, Zeeshan Zakaria zisha...@gmail.com wrote:
Do you recommend using wav files instead? Will there be any
Are you using app_swift or wav files?
On Oct 23, 2010, at 5:26 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Hello list,
I have been using Cepstral's 8KHz voices for my text-to-speech service for
some time now, and have been noticing that the voice quality is really poor,
doesn't
We recently completed a project using products from here:
http://www.controlbyweb.com/webrelay/ They were easy to setup and can
be controlled in a variety of fashions included http queries.
Darren Wiebe
On 18/10/2010 8:34 AM, Marco Signorini wrote:
Hi
Did you looked at Arduino + Ethernet
Just thought I'd let everyone know I've got a new beta version of app_swift up
for Asterisk 1.8 on http://forge.asterisk.org.
- Darren
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On Sep 18, 2010, at 11:41 AM, Mark Deneen wrote:
On Fri, Sep 17, 2010 at 11:52 PM, Dean Collins d...@cognation.net wrote:
Any thoughts on why the lack of traffic?
Cheers,
Dean
Not enough applications to play immature bathroom sounds.
You could well be right, but consider for a
or feedback, please let me know.
Thanks,
- Darren
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when you're done. You can
also disconnect calls from the asterisk cli using the soft hangup command.
Darren Wiebe
dar...@aleph-com.net
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server is an Eloqua box,
... that's the CRM technology Digium uses to track their campaigns.
-Darren
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file and
plays back the user name recording. Hasn't added any CPU overhead with the call
processing and along with working as intended I think there maybe some other
unique capabilities for it down the road.
In any case, thought I'd update the thread.
Cheers,
- Darren
On Jan 11, 2010, at 10
as designed (DTMF detection, etc.).
Any ideas or help would be appreciated.
Many thanks,
- Darren
POI:
Asterisk 1.6.1.6
app_meetme.c - line 1601 (the announce_thread function)
app_meetme.c - line 1817 (the conf_run function)
-- snip --
#!/usr/bin/perl -w
use strict;
use warnings;
use lib
) or M0n0wall. I've had good luck
with both of those.
Darren Wiebe
dar...@aleph-com.net
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asterisk
.
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dar...@aleph-com.net
Aleph Communications
www.aleph-com.net
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purchased a few copies of it is that
I need to have several different sip and iax2 connections for testing
purposes.
--
Darren Wiebe
dar...@aleph-com.net
Aleph Communications
www.aleph-com.net
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My thoughts were similar. Availability has not been a problem for us on
the WRT54GL boxes. We're pulling them out of our wholesaler all the
time without any problems.
Darren Wiebe
dar...@aleph-com.net
Jeff LaCoursiere wrote:
And why not DD-WRT, which runs on many more platforms including
Just restarting it won't do anything. You could use the following
command to find any files over 200mb on the system. Be careful about
blindly deleting stuff though
*find / -type f -size +200M
Darren Wiebe
dar...@aleph-com.net
*
David @ULC wrote:
I have 320 GB SATA HDD.
When I
be happy to try it again to see if I've become a male yet. :)
Darren Wiebe
dar...@aleph-com.net
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sorry to but in, but...
1 on first line make sure it is 3!/usr/perl/bin not
#!/user/perl/bin
I'd suggest instead: #!/usr/bin/perl
;)
2009/2/21 Yawar Hadi yawarh...@gmail.com
hi steve,
plz make some cahnges and now i have tested it its working fine to me
1 on first line make sure
] logger.c: -- Hungup 'IAX2/ToHK1-16'
This tells me when the call was terminated, but doesn't tell me which party
actually hung up first.
Is this possible to detect?
thanks,
Darren
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Pretty cool. I'm almost offended though as I'm not usually guessed as a
female of the species. :)
Darren Wiebe
dar...@aleph-com.net
Asterisk Asterisk wrote:
Steve,
Tried to test and got call could not be completed as dialed.
Were you able to connect? If not, please try again. Call volume
We've done this with good results. You can also get one that flashes a
bright light for not a lot of money.
Darren Wiebe
dar...@aleph-com.net
Steve Gladden wrote:
If you wanna go low tech. down dirty you could also go with a conventional
POTS phone line 'loud ringer' device and simply hook
* - this
is your first test).
Sincerely,
--
Darren Nickerson
Telephony Depot
www.telephonydepot.com
+1.215.825.8710 ext 8106 (office)
+1.215.243.8335 (fax)
* http://www.fiercevoip.com/story/skype-voip-dead/2008-09-17
* http://voxilla.com/2009/01/19/is-the-2009-voip-surge-theory-correct
What version of Asterisk and what version of app_swift?
On 29 Oct 2008, at 15:10, [EMAIL PROTECTED] wrote:
Hi, I have tried installing app_swift on both mac os x and ubuntu now
and am getting the same error. I must be missing something, as I have
tried multiple versions and everytime do sudo
do not need to link to Asterisk, etc.
Darren Wiebe
[EMAIL PROTECTED]
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Not sure if this counts as affordable but:
http://www.voipsupply.com/cyberdata-voip-intercom
-Darren
On Mon, Oct 27, 2008 at 8:46 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Mon, Oct 27, 2008 at 8:36 AM, hbk [EMAIL PROTECTED] wrote:
Hi,
Is there an affordable HW solution to do a door
Not sure about the Swedish, but Lumenvox has a great speech
recognition app for Asterisk.
- D
On 26 Oct 2008, at 19:53, Christian wrote:
Hi all,
Yes, this might not be the proper list for this, but i have a
question about Asterisk and voice recognition.
If I want to create a menu
thanks,
Christian
On 2008-10-26 at 20:32 Darren Sessions wrote:
Not sure about the Swedish, but Lumenvox has a great speech
recognition app for Asterisk.
- D
On 26 Oct 2008, at 19:53, Christian wrote:
Hi all,
Yes, this might not be the proper list for this, but i have a
question about
thereof) - and this has been somewhat fruitful, if not quite
tedious.
It would be nice to have a reference guide that lists the most common
log messages, and what they mean.
Does such a guide exist?
thanks,
Darren
--
DOCOMO interTouch provides a full suite of integrated solutions
Well, after very quickly making a test call it's not Vitelity. It could be
something with your account? Might want to try opening a support ticket. If
you want, create a sub account and e-mail me off list the username and
password and I'll test it with my box or vice versa.
On Tue, Oct 7, 2008 at
Interesting, I've been using them since April and haven't had a problem. I
know they changed their server settings a while back but didn't notice
anything recently.
On Tue, Oct 7, 2008 at 11:47 AM, Roderick A. Anderson [EMAIL PROTECTED]wrote:
Darren Severino wrote:
Well, after very quickly
)
That equates to goto the default context, extension 101, at the 1st
priority which is your Dial command.
Best Regards,Darren Severino
On Sat, Oct 4, 2008 at 1:30 PM, Stephen Reese [EMAIL PROTECTED] wrote:
I have a Asterisk server setup and I am able to connect to the server
using a soft client
actually wrote one of these ages ago that worked fairly well with
a10 calls per second SER server. How many calls per second are you
looking to process?
- D
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
I know. :)
I've already mentioned some of the OpenSIPS options to him on the
OpenSIPS users list (LCR module specifically). Just brain dumping
everything that came to mind.
- D
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
with OpenSER for
such a small amount of users.
Asterisk can do everything you'll need it to do otherwise.
- D
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Oct 1, 2008, at 7:44 PM, Alex Balashov wrote
Any particular reason you're using H323 instead of SIP ?
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Sep 16, 2008, at 12:04 PM, Guilherme Loch Waltrick Góes wrote:
I have a Cisco 3845 with a ISDN PRI port
this type of general setup in the past with a
great deal of success for remote offices and soft-phones on laptops.
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Sep 9, 2008, at 1:19 PM, Mattias Andersson wrote
You shouldn't have any delays at all.
Are you using ztdummy for timing? and what kind of load does the box
have on it?
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Sep 9, 2008, at 4:23 PM, George
A cheaper alternative would be the voip wiki.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Sep 4, 2008, at 12:13 PM, Mark
Impressive work Bradley! I tested it and it worked great, even with my
mandatory 'use strict'.
Thanks,
- Darren
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 29, 2008, at 5:47 AM, Watkins, Bradley
handset:
http://www.telephonydepot.com/product_p/105-058-8002dual.htm
One limitation is that there's no minibrowser, so you won't be able to
navigate the http proxy signup/authentication page at your local
coffee shop. Works great in the typical office setting though!
Sincerely,
--
Darren
-verbose(”No subroutine name passed!!”, 1);
return(-1);
}
my $exec = \{$sub};
return($exec-());
}
set_variables();
dynamic_execute(”run_me”);
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
smime.p7s
Description
Are you using an Asterisk PBX?
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 27, 2008, at 7:06 PM, Tom Moore wrote:
Hi guys,
What are your suggestions to people who have pbx systems that
interface
You can use an extremely simple Asterisk config to do the SIP-PRI
call conversion that'd be very solid.
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 27, 2008, at 7:37 PM, Tom Moore wrote
For simple paging the bogen tamb works very well. Just hook it up to an
fxs port and you're good to go.
Darren Wiebe
[EMAIL PROTECTED]
Jonathan Disher wrote:
I am looking to replace the phone system at my father's shop with an
Asterisk box and some Cisco phones, but one piece
, then that would also explain why outbound PSTN DTMF is
functional.
Hope this helps.
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 23, 2008, at 12:39 AM, Max Alex wrote:
Hi everybody,
I have linksys phone
satellite.
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 23, 2008, at 4:45 PM, Femi wrote:
I’ve used VOIP over satellite for years and while it’s not perfect
it is sometimes actually better than cellular
Just change your dial command and add the plus sign there.
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 22, 2008, at 1:28 AM, ronald wrote:
Hi,
Is it possible to assign a plus sign on the callerid(num
, and the problems lies somewhere on your network between the
Asterisk server and whatever gateway / device. If it sounds awful, and
the codecs match, then it's time to start troubleshooting the server.
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
Not sure what you've heard before, but I have successfully used a
modem at 9600 baud (forced via AT commands) through a zaptel card on
several occasions.
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug
We recently discussed DeadAGI on the list - I'd check the archives
first.
I just finished doing a write up on DeadAGI and Perl on my website if
you're interested.
DeadAGI *can* be very reliable if done properly.
- Darren
_
[EMAIL PROTECTED]
http
I'd run top on the server to see if the CPU utilization is going
through the roof. If you use AGI, make sure there aren't any orphaned
processes consuming resources.
If all else fails on the software side of things, I'd restart the
server.
_
Darren Sessions
';
$SIG{QUIT} = 'cleanup';
$SIG{HUP} = IGNORE;
With this approach, as I said before, I've ran perl agi apps in very
high call volumes at various companies for years without any issues.
Hope this helps.
- Darren
_
[EMAIL PROTECTED]
http://www.darrensessions.com
to see
what kind of utilization your local server is at just to make sure
something isn't wrong there either.
Hope this helps,
- Darren
_
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 18, 2008, at 10:41 AM, Nikhil Nair
.
I just posted a Perl based call file generator to the list not to long
ago that would easily work for this application.
Hope that helps,
- Darren
_
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 13, 2008, at 4:20 PM
with
whatever parameters to create as many call files as you felt like, and
Asterisk would start acting on them immediately (if the call files
were generated without wait time).
- Darren
_
[EMAIL PROTECTED]
http://www.darrensessions.com
http://www.linkedin.com
.
Hope this helps,
- Darren
_
[EMAIL PROTECTED]
http://www.darrensessions.com
http://www.linkedin.com/in/dsessions
_
On Aug 6, 2008, at 7:02 AM, Guilherme Loch Waltrick Góes wrote:
I have a server with Asterisk 1.4.21.1 and some
.
See if that does anything for you.
- Darren
_
[EMAIL PROTECTED]
http://www.darrensessions.com
http://www.linkedin.com/in/dsessions
_
On Aug 6, 2008, at 8:48 AM, Guilherme Loch Waltrick Góes wrote:
I'm using OpenSUSE 10.3, the funny
I have used virtually all versions of Asterisk 1.0+ (literally, either
in production or testing) with OpenSUSE 10+ and 11 on AMD and Intel
and haven't had any issues with gcc optimizations with regards to
audio sounding choppy. This scenario for me has always been the gsm
libs.
ASTPP (www.astpp.org) will do calling cards / prepaids as well as lcr.
Darren Wiebe
[EMAIL PROTECTED]
emist wrote:
Hello,
does anyone know of a good calling card solution for asterisk that is
able to do lcr?
Does astcc do this? I've been searching around and I can find some lcr
modules
on
a separate set of boxes.
I'm not exaggerating when I say the replication was up and running in
about 10 minutes.
While I do appreciate (a lot) how standards compliant Postgres is,
MySQL was an absolute clear winner in my book with regards to the
replication.
Just my two cents . .
- Darren
Additional details are in the ChangeLog and README files in the tar
ball.
As always, if there are any questions or comments, please forward them
to me at [EMAIL PROTECTED]
Thanks,
- Darren
_
[EMAIL PROTECTED]
http://www.darrensessions.com
http://www.linkedin.com
If you had a dax in front of all your circuits, you could move them
from one server to another without physically touching anything.
I've done about 300 calls on a dual processor box doing just SIP with
an entirely AGI based setup and it held up just fine, but doing TDM,
I'd worry about
I had issues like this on one installation that cleared up when I turned
ACPI and APIC?? off in bios.
Darren Wiebe
[EMAIL PROTECTED]
Gerard A. Matthew wrote:
Are your phones behind NAT?
This should be an issue with rtp port communication.
Gerard.
--Original Message--
From: John
args are specified
Can now wait for DTMF after text-to-speech processing is done if the
timeout and max digits args are specified
Entire DTMF input placed into channel variable
Can be downloaded from http://www.darrensessions.com
Thanks,
- Darren
,
- Darren
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-base version is almost complete.
Thanks,
- Darren
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Are you trying ethernet or serial? Have you tried the other?
-Darren
From: [EMAIL PROTECTED] on behalf of C F
Sent: Thu 5/22/2008 1:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Adit 600 password reset
Just FYI, I wrote an application that tracks the status of SIP or IAX2
extensions by listening to the AMI. It was for use by callshops but
would probably require minimal change to work for you. It's currently
part of the ASTPP source code.
Darren Wiebe
[EMAIL PROTECTED]
Atis Lezdins wrote
If you're willing to cc me a copy I'll be in your debt.
Thanks,
Darren Wiebe
[EMAIL PROTECTED]
Steve Totaro wrote:
On Mon, May 5, 2008 at 5:10 PM, Roderick A. Anderson [EMAIL PROTECTED]
wrote:
Steve Totaro wrote:
On Sun, May 4, 2008 at 1:55 PM, Roderick A. Anderson [EMAIL PROTECTED
Am I correct in thinking that one application of this would be
monitoring what you have left for funds with a prepaid vendor?
Darren Wiebe
[EMAIL PROTECTED]
Brian J. Murrell wrote:
On Wed, 2008-04-23 at 09:38 -0700, Nhadie Ramos wrote:
Hi, sorry to confused you with my question
Ok, I'm not aware of this feature in astcc and I can't speak for astbill
or a2billing. I do know that I coded it into astpp and it's called
vendor rating in there. It works but it's not used a lot at present.
Darren Wiebe
[EMAIL PROTECTED]
Nhadie Ramos wrote:
hi sir,
yes that would
first release of anything.
Thanks,
- Darren
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I've used Adit600's almost exclusively for my installs. All have worked great
for me.
-D
From: [EMAIL PROTECTED] on behalf of Steve Totaro
Sent: Thu 4/3/2008 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
CentPBX has bit the dust I believe.
-D
From: [EMAIL PROTECTED] on behalf of Chris Bagnall
Sent: Wed 4/2/2008 7:12 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] CentPBX mirror?
Greetings list,
Not exclusively
Red Hat, Suse, Debian, ?
Was there some key feature it offered that the others didn't ?
Cost ?
Darren Wright wrote:
Notifications can be done either thru SNMP traps or SMTP. Insight
Manager is free from HP, but any SNMP trapper can work with alerts.
The recovery CD is just a build that reloads
controllers and like 20 CPUs to chose from. How did
you
chose ?
Thx for sharing !!!
Darren Wright wrote:
One of the major reasons we use DL320 / DL380's is the ease of
swapping
drives, and the integrated ILO BIOS level access.We can support
remote
sites with ease.
If a drive dies we
One of the major reasons we use DL320 / DL380's is the ease of swapping drives,
and the integrated ILO BIOS level access.We can support remote sites with
ease.
If a drive dies we get a notification, a new one is sent and a non-techie can
replace it with guidance.No onsite visit.
=asreceived
faxdetect=incoming
nsf=sdn
group=1
channel=1-23
zaptel.conf
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
On 3/15/08, Darren Wright [EMAIL PROTECTED] wrote
Feel free to ping me off list. I've setup quite a few Cavtel PRI's with *.the
paperwork they asked you to setup?
Typically, they only send 4 digits.
Do you have the questionnare they asked you to fill out?
dwright at d2 - tech dot com.
From: [EMAIL
been
released to distributors yet. I think we're still a couple of weeks out on
that product.
Sincerely,
--
Darren Nickerson
Senior Sales Support Engineer
Telephony Depot
www.telephonydepot.com
+1.215.825.8710 ext 8106 (office)
+1.215.243.8335 (fax
You'll want to use the XML park and pickup with the aastras.
Feel free to ping me off list if you need help.
-Darren
Dwright at d2-tech dot com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of OCG
Technical Support
Sent: Monday, March
I've used lots of Digium T1 cards on DL380 / DL320's without a hiccup.
-Darren
From: [EMAIL PROTECTED] on behalf of Joshua Kinard
Sent: Tue 2/26/2008 5:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Had
Yup, SIP is working ok as well, except for the cross-country 100ms round trip.
Their answer was to upgrade to 1.4
Not an option for me.
Please ping me off list so we can further discuss.
dwright at d2 - tech dot com
-Darren
From: [EMAIL PROTECTED
, but no flat charges, no 729.
Frankly, even my broadvoice (yikes!) connection has been significantly
better, no 729.
For a full Virutal PRI, I'd look at a provider that can give you the
port and SIP connections, like XO. I've had good success with XO's
product.
-Darren
The HWEC, not software.
-Darren
From: [EMAIL PROTECTED] on behalf of Olivier
Sent: Thu 2/21/2008 12:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Which echo-can for Digium B410P ?
Hi,
Which echo-canceler shall
An analog DID trunk is a line (typically part of a group) that has a group of numbers assigned to it at the telco side. They work in a variety of ways depending on the telco. One example is the trunks as Telus provides them. The end user provides dialtone back to the telco. When a call comes in
this feature. Okay so I will take the lead and pimpit for asterisk. With Rhino Analog cards you CAN do ADID with no extraequipment. However if you want to spend the money we can go the otherroute :)darren wrote: An analog DID trunk is a line (typically part of a group) that has a group of numbers assigned
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