On 2023-05-23 7:22 p.m., Steve Matzura wrote:
And I think they're both small.
[May 23 18:34:12] NOTICE[46582]: res_pjsip_session.c:3968 new_invite:
voipms: Call (UDP:208.100.60.12:5060) to extension 's' rejected because
extension not found in context 'voipms-inbound'.
Steve,
In your
On 3/25/2010 8:13 AM, David Gibbons wrote:
Hi All
I'm involved in discussions with my carrier right now and am wondering if
anyone has interconnected Asterisk to Metasphere via SIP?
Yes, we're served by a Metaswitch usng SIP. Works fine.
-Daryl
IAX calls are
being passed on to a far end terminator via SIP.
I was going to scrap IAX entirely because it didn't seem to scale well
(for non-trunking apps, at least), but many customers need it for
various reasons.
Daryl
On Nov 30, 2007, at 8:52 AM, zoa wrote:
IAX had some stability
(CALLERID(num)=${IF(${REGEX(^(?:\([2-9]\d{2}\)\ ?|[2-9]\d{2}(?:
\-?|\ ?))[2-9]\d{2}[- ]?\d{4}$ ${CALLERID(num)})}?${CALLERID
(num)}:staticNumber)
I'm sure I'm pretty far off - and I've been through many permutations
of this so far. Any ideas?
Thanks,
Daryl
of whether a play progress tones or not
from the AGI.
Daryl
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that will automagically happen
whether I want it to or not? If so, I'm going to have to do some
ugly dial plan scripting to make this work.
In case it matters, this is a PHP AGI.
Thanks,
Daryl
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I'm having a problem building Asterisk 1.2.22. It fails in
codecs/codec_zap.c on codec_zap.c is revision 62173. The OS is FC4.
Here's the error. Can anyone help me with this?
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT
are using an older one.
on Wednesday 07/18/2007 Daryl Jones([EMAIL PROTECTED]) wrote
I'm having a problem building Asterisk 1.2.22. It fails in
codecs/codec_zap.c on codec_zap.c is revision 62173. The OS is FC4.
Here's the error. Can anyone help me with this?
gcc -pipe -Wall
That's what I needed to know. Thanks!
John covici wrote:
But asterisk will not compile till you install the correct version of
zaptel.
on Wednesday 07/18/2007 Daryl Jones([EMAIL PROTECTED]) wrote
Correct. zaptel-1.2.12 is currently installed. I plan to install
zaptel-1.2.19 as part
Bill Michaelson wrote:
Is it just me, or is the AGI interface at cnam.got-name.com failing
for others? Anyone know how to contact them without sending postal
mail or telegram?
I don't know how to contact them, but I am having the same problem.
I've fired a script from an AGI-BIN to accomplish that.
Try this one:
#!/usr/bin/perl
# mk 2004 feel free to distribute
# [EMAIL PROTECTED], _Vile
# perl script to reboot phones
# try telnetting to your phone, first.
#
use Net::Telnet ();
$phone_ip = shift;
# Your Cisco 79xx prompt
On May 14, 2007, at 11:27 PM, Atlanticnynex wrote:
I'm curious what kind of configuration/features/modules you could
recommend for my setup. Can you explain further what you mean by
OpenSER to Asterisk?
If you want to go Open Source, I think OpenSER is a good choice. You
won't need to
On May 12, 2007, at 4:11 PM, Atlanticnynex wrote:
Thanks Alex, some great ideas.
I think, however, I'm leaning towards Asterisk at this point- since
I have quite a bit of experience there, and very little with SER.
At this point, I'm wondering from a dimensioning standpoint, what
kind of
On May 14, 2007, at 1:29 PM, Zoa wrote:
Several people do use it for handling 50k minutes a day. (I'm one
of them).
Yes, you need to know what you are doing, and have a nice design,
but it is possible.Our code is only slightly altered. (mainly for
billing purposes).
That's great if
successfully (re)INVITED and the media was no
longer going through my Asterisk box, but ethereal says different.
I'm not having much luck finding any information on this on the wiki
or google. Can someone point me in the right direction?
Thanks,
Daryl
given time). I've been going the ethereal route, which
is great for debugging, but not so good for a quick look.
Thanks again,
Daryl
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[16526]: chan_zap.c:6184 ss_thread: Got event 18 (Ring
Begin)...
-- Executing DISA(Zap/1-1, no-password|net_outgoing) in new stack
--
Daryl Sayers Direct: +612 95525510
Corinthian Engineering Office: +612 95525500
Suite 54, Jones Bay Wharf
I'm having trouble with Cisco 7960 and Linksys SPA-942 SIP phones not
displaying the Caller-ID number. The Caller-ID name is displayed, but
not the number. Instead, the phones always display the value that's set
in the fromuser= parameter in sip.conf. If fromuser= is not set, then
the
not work. The calling number and name are both
properly displayed on all of the softphone clients that I've tried.
Here's the format I'm using to set the CallerID.
SET CALLERID JONES DARYL A6508701826
Can anyone help?
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Tim,
I have seen the same 400 errors and the broken MWI... I backed up to
7.3... We'll see if Cisco corrects these in the next release...
Daryl
- Original Message -
From: Tim Connolly [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
I've been using the following macro to ring SIP and IAX devices for a
few seconds, and then add on a cell phone if there is no answer on the
SIP or IAX device. Periodic problems began a few versions ago and now
the problem happens every time with 1.2.9 and 1.2.9.1.
The problem is that when a
The @ip-address is actually a documented cisco fix to another problem.
I'd have to look it up, cause I don't remember exactly what it was, but
it's been on the list somewhere, and I think EVERYONE that's used 8.2 has
the same problem with the firmware. I would suggest using 7.4 or 7.5.
ancellation on channel 1 Jan 19 15:18:05
DEBUG[24046] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Jan
19 15:18:05 DEBUG[24046] chan_zap.c: Updated conferencing on 1, with 0
conference users Jan 19 15:18:05 VERBOSE[24046] logger.c: -- Hungup
'Zap/1-1' Thanks,
Daryl
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and I can place outbound calls. The
only issue I have is sending inbound calls to the Cisco device. Any
thoughts???
Daryl
- Original Message -
From: Adam Rybak [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users
direction?
Thanks for the help,
Daryl
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It's not just you. Same thing happens here. I went back to 1.0.7.
Stefan Gofferje wrote:
Hi folks,
I used to have some constructions like
exten = number/callerid,1,Goto(somewhere)
After updating to 1.0.8 those does not work any more.
Any hints?
Regards,
Stefan
in it and it supports a
very small office with mixed SIP and POTS inbound/outbound. Running
Debian, of course.
Daryl
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to (in the US anyway) certification as intrinsically
safe.
I don't know either way about phones listed as such, but with the right
terminology you might have better liuck searching.
voiceverified. | Daryl G. Jurbala
-- | Chief Technology Officer
| 215.862.1160
.
voiceverified. | Daryl G. Jurbala
-- | Chief Technology Officer
| 215.862.1160 x235 (Office)
It had to be you! | 215.862.9880 (FAX)
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are in or very near a city you can, but not
everywhere.
You find me a reliable Teir 1 ISP T1 in New Hope, PA for $300 to $400
and I'll give you the amount I save over the next quarter. NPA-NXX is
215-862. Good luck.
voiceverified. | Daryl G. Jurbala
-- | Chief Technology
people who do things like this just never consider the life
safety risk involved until its way too late.
I'll get off my soap-box now and get back on topic.
Daryl
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in them. If
you run this setup well past the time it was designed to run (by adding
3, 4, or more times that battery capacity it was ever designed to have)
that chances of a catastrophic inverter failure (meaning flash, boom,
fire) are very real and very likely.
Daryl
help?
Just a crazy idea herehave you contacted NuFone support yet?
Daryl
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).
Unless DNIS has turned into telcoBGP while I haven't been watching, what
you're being asked to do doesn't seem quite right.
Daryl
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Has anyone had any experience (good or bad) running Asterisk under
VMWare ESX server on a blade chassis? This application will (fairly
obviously) not include Zap channelsactually, it will be SIP-only.
Please feel free to contact me off-list and I'll summarize for the list
later.
Daryl G
encapsulated VPN would have QOS, the underlying
transport medium (IPv4) still would not (if it didn't have it before).
Furthermore, if any Ipv4 hops in between would have prioritized your
traffic higher based on its type, they now have no idea what is is,
because it's encapsulated.
Daryl
professionally when
reporting the issue and received the same treatment in return. The
issues were also resolved promptly.
Daryl
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the interface happen. It's
like asking what card you need to connect your computer to some
undescribed network. If the network is ethernet, you need an ethernet
card. If it's token ring, you need a token ring card, etc.
Daryl
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.
The Dialogic IVR SDK monitors call termination status this way, so I'm
looking for something similar in *. Anyone have any ideas on this one?
Or am I going about this the hard way and missing an obvious
alternative?
Thanks,
Daryl
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Yes, I'm replying to my own post.
Roger Gulbranson suggested this:
http://www.voip-info.org/tiki-index.php?page=NVBackgroundDetect
As he's using it for FAX detect, and it has a talk option as well.
If anyone is interested, I'll report back with my results.
Thanks Roger!
Daryl
have a question to ask, I'll call you
first to ask for what search terms to use before posting.
What's your mobile number?
Thanks,
Daryl
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.
[...]
Not to just me too, butme too. I've contacted their support on
numerous occasions, and have been given busywork to do (run ping plotter
for 24 hours, send us the results, etc) and never receive a response
that acknowledges a problem of any sort.
Daryl
to all who helped,
Daryl
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PLEASE CONFIGURE YOUR
AUTORESPONSERS TO NOT SEND MESSAGES TO PEOPLE WHO POSTIN MAILING LISTS YOU
SUBSCRIBE TO.
This is an extremely rude
thing to allow, and is becoming increasingly common, especially with users of
the Asterisk-Users list.
Daryl
From: [EMAIL PROTECTED]
[mailto
early after coming
off-hook because I can just redial and have it work (or not) randomly.
Does anyone know what this might be and/or an easy way to have the ZAP
channel come off-hook, delay for 1/2 second or so, and then dial?
Thanks,
Daryl G. Jurbala
NGM Tec, Inc.
Tel: 215-862-1160 ext. 235
Fax: 215
long
distance and no metered minutes for about $37 a month. A BRI costs you
about double that for the loop, with metered minutes and bring your own
LD.
Past the technology aspects, BRI just doesn't work here. And I'm going
to guess that pricing structure is similar in other areas as well.
Daryl
has to make excuses for it (sorry..it's VoIP).
Anyone who wants to run junk hardware and beta code pretty much loses
their right to complain about the results of doing so.
Daryl
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is. Minutes handled
by bold old and new companies.
Now if you wanted to say that it's not in vogue for soft PBXen and key
systems to support h.323, I'll buy that. But I'm going to guess that
voice traffic over SIP is a mere fraction of voice traffic over h.323 on
any given day.
Daryl Jurbala
Brent Franks wrote:
I have some reports from users that occasionally DTMF will stop working in
voicemail and they will have to exit the system to get it to work again.
The useragents are Polycom IP 500's and I am using dtmfmode=rfc2833, with
Ulaw codec. This is all on an internal switched 100mb
Nope...I scrapped that idea and just bought a Digium card.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of greg
Sent: Thursday, July 22, 2004 2:08 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk and Linejacks
I found a message from
(and by junk I mean anything you can buy at Staples that claims
to be a router).
Daryl
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.
None of it struck me as particularly professional.
If there is someone from Voicepulse here, feel free to stand up for
youself and tell us your side of things. From here it's not looking too
good.
Daryl
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you choose on
1.) failure of their systems or 2.) loss of connectivity to the remote,
no matter who's fault it is. This should be automatic, seamless, and
the forwarding number should be changeable by the account holder on the
fly.
Daryl G. Jurbala
BMPC Network Operations
Tel (PA): +1 215 825
termination or origination working, plus this message made me simply
skim it.
Anyhoo, it looks like the old method isn't working at all anyway.
Switching to the new method for termination worksbut I still don't
have origination, and I've still waiting on hold (27 minutes and
counting).
Daryl
origination takes a bit of priority over their choice
of version.
FYI, origination magically started working aging about 45 minutes ago.
Daryl
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! Maybe it will even keep working if x doesn't decide to
change it or start charging me, etc.
For now, that leaves people in my position paying for PRIs or POTS lines
just to be sure.
Daryl
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I'm having a problem with intermittent lookup of Caller ID Name info
using LookupCIDName.
The same problem occurs when doing:
asterisk -rx database show cidname
No data is returned on every fourth or fifth query. No errors are being
logged.
I'm currently running CVS-HEAD-07/07/04-17:04:31 and
I've been reading drafts of this book for at least nine months and can
assure you that its content is very different than what is available on
the Wiki. The book is an excellent introduction to VOIP in general, and
offers sufficient information for the novice to configure a basic
Asterisk
me (nicely or not...I
really don't care).
Daryl G. Jurbala
BMPC Network Operations
Tel (PA): +1 215 825 2107 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 215 862 9880
INOC-DBA: 26412*DGJ
PGP Key: http://www.introspect.net/pgp
Actually, if all of the outside lines are full than ca just get a
reorder tone for all it mattersbut yes, basically 96 desk stations
is what we're talking about.
Thanks for the pointer. I'll look into the Adits. Certainly sounds
like the price is right.
Daryl
-Original Message
-7206VXR-DS-3-7206VXR introduces only 12 ms latency on
average. Of course that's nearly $30k worth of plumbing, so one would
expect that kind of performance.
Daryl
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And it actually is.the only problem is that the downloads on the
Cisco site are actually CallManager updates. So you'd need a CM server
to extract the image file (which you could then toss on whatever tftp
server you want).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
The default password is Admin.
Adam Goryachev wrote:
On Tue, 2004-04-06 at 14:36, Christian Hoffmeyer wrote:
Thank you for all of the beta testing. New and improved graphics in this
release along with
drag and drop transfers and hold for all technologies.
There's a screenshot on the link
config.
Daryl
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I contracted with Digium for this enhancement and am waiting for it to be
completed.
Tilghman Lesher wrote:
On 2004 Apr 02, at 12:04, Brian Capouch wrote:
I don't want to re-invent the wheel if someone has already
hacked a way to do this.
One of my customers has a number of stores, and he
What you and so may others on this lise seem to forget is that Cisco is a company
offering bsuiness products for businesses. Businesses typically pay by check and wire
transfer, especially for items such as this.
If you want home-user pay-by-credit-card service, buy products from Belkin's
different NAT
boxes. Canreinvite=no will force your home phone to always pass its
traffic through the * box, eliminating the issue you are having.
Daryl
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these are the same phones? Because I still have
boxes of them somewhere too (that seems to be a common thread here).
Daryl
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800 numbers all the time from my Nufone account
without problem.
Hell, my DID through Nufone -IS- an 800 number!
[...]
Of course you're aware that a DID and call termination are completely
different things that have nearly nothing in common.
Daryl
argument). This
one is about Asterisk. Not the services you can use Asterisk with.
Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ
...
Is it possible to use the asterisk to initiate a call to a phone?
Yep...it's on The Wiki at http://www.voip-info.org.
Specifically, I think
http://www.voip-info.org/wiki-Asterisk+auto-dial+out is what you're
looking for.
Daryl
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working (or not working)
baords would be highly appreciated.
That's not a PCI Expansion Slot. It's a passive backplane, designed
to host a single-board industrial type machine.
Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208
,
Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ
PGP Key: http://www.introspect.net/pgp
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[EMAIL
actually SSHed
into my box and fixed it/showed me what I was doing wrong.
The support is well worth the price, especially if you are building a
production server. Or if your time is worth anything at all for that
matter.
Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI
on a 7960. Crank up the packets
and try to make phone calls. Then we'll talk again.
Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ
PGP Key: http://www.introspect.net
.
And if you also have a Cisco wireless infrastructure (AiroNet 350 and
newer) you can power those with the same hardware.
Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ
of this train.
Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ
PGP Key: http://www.introspect.net/pgp
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their own
implementation, it makes the standard not so standard anymore.
Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ
PGP Key: http://www.introspect.net/pgp
control methods made much more accessible with SIP
phones and large, programmable displays with soft buttons.
Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ
PGP Key: http
it was working under all conditions. Personal experience shows
that it is most definitely not on Cisco and 3Com products. Others have
told me their stories with other manufacturer's equipment. None of it
was good.
It's not a production-stable way to deploy phones. Period.
Daryl G. Jurbala
BMPC
directory for
each topic consisting of symbolic links to the real sound files.
That's how I currently handle things on my systems.
Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 508 526 8500
INOC-DBA
gone wrong, you need to know what you're doing. Its your
job (the people I'm talking about know who they are), so do as you wish,
but I sure would install something I know little or nothing about and
call it production.
Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI
a capability standpoint.
Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ
PGP Key: http://www.introspect.net/pgp
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for collisions,
etc. on the port.
Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ
PGP Key: http://www.introspect.net/pgp
knots is a speed unit (like
Miles Per Hour), so I think you want knots here, not knots
per hour, if you are talking wind speed.
[...]
Then stick to being a land lubber. Because you're wrong.
A knot is a unit of linear measurement.
Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401
wrong or right :)
http://www.yourdictionary.com/ahd/k/k0092800.html
[...]
Truein common usage a knot means either nautical mile or nautical
miles per hour depending on the context.
That was a pre-coffee post. I can take only partial responsibility. ;)
Daryl G. Jurbala
BMPC Network
as to the feelings of the regular
contributors and answer-providers on this issue?
Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ
PGP Key: http://www.introspect.net/pgp
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or super lists. My guess is that
there are other
MLMs out there that have similar features.
LISTSERV is evil, and yes, there are (listserv is evil mostly because of
the abhorrent cost of something that is available via open source/free
alternatives and a couple of perl/awk/sed scripts).
Daryl
relative.
The problem with splitting VoIP and T1/TDM/whatever you want to call it
is that the crossover is huge, and where the problems lie often aren't
clear to those looking for help.
Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ
PGP Key
of a proper commercial predictive dialer would be
relatively cheap after already having sold your soul.
Daryl
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,
or are you having a problem possibly with the asterisk box? Some other
things to know: are you running voicemail yet? If so and you can dial
into it from either of the phones, how does it sound? If not, how about
anything from the * boxlike the demo annoucment stuff?
Daryl
providing test boxes hosted at on of my NOCs. Chime in if
you're interested ([EMAIL PROTECTED]).
Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ
PGP Key: http://www.introspect.net/pgp
you suggest, only one of
them will ring. It appears to me that the one that is fastest to
respond will work, but I only tried the setup briefly before doing a bit
of research that told me it wasn't the way this is done in *.
Daryl
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quality to me.
Daryl
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a mouse or
keyboard if properly configured. Most hardware can't even detect if a
monitor is attached or not.
Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ
PGP Key: http://www.introspect.net/pgp
scenarios are valid in the real world.
Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ
PGP Key: http://www.introspect.net/pgp
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many people on this forum will be
more than happy to assist you in getting * to use a properly functioning
mailer to send voicemail notification.
Here's more unsolicited forum posting advice: the attitude you are
taking in the above quoted post does not inspire people to help you in
any way.
Daryl
so.
[...]
Steven Critchfield [EMAIL PROTECTED]
Steven...where can I find this script, or can you forward a copy to me?
Sounds like exactly what I need at this point.
Thanks,
Daryl
[EMAIL PROTECTED]
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support costs money. Example: Digium.
Daryl
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/sound card?
I only want to use it as a IP PBX.
YES
you can.
how about IAX2 trunking? does this work with ztdummy?
I was using both IAX2 trunking and MOH before getting my zap devices,
and I never had any luck with ztdummy.
Daryl
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