On 02/14/2011 12:04 PM, James Miller wrote:
I did the command listed, and its actually requesting RINGLIST.DAT, so I
changed the filename to match its request but now its showing in the
ring type setting:
Chirp 1
Chirp 2
24 24-ring-tone-1.raw
Att1 ring_att1.pcm
snip
Do you actually have
On 02/07/2011 11:46 AM, Gilles wrote:
snip
Asterisk runs as root, and owns this file as well.
Have you tried setting the permissions of this file to world readable,
to ensure that any user can read it and eliminate potential permissions
problems?
Worth a shot. While you're at it, output
On Feb 3, 2011, at 11:12 AM, Kevin P. Fleming wrote:
The Queue() application can automatically pause members who fail to answer;
this would be the solution to your problem. With that solution in place,
though, the agent will still need to be able to un-pause when they return to
their
On 01/31/2011 12:51 PM, salaheddine elharit wrote:
I have asterisk installed in our call center and i want to know how to
do in order to save all the calls (inbound and outbound) if there is any
tool
Yes, there is.
Tom
PS: Sorry, I couldn't resist!
--
On 01/31/2011 12:51 PM, salaheddine elharit wrote:
I have asterisk installed in our call center and i want to know how to
do in order to save all the calls (inbound and outbound) if there is any
tool
OK, now to be somewhat more helpful, this is a common scenario. You
should search for
On Jan 30, 2011, at 4:21 AM, Pezhman Lali wrote:
Dear,
Faxter is an opensource email to fax gateway,
please check it, let me know if any bug.
best
I'll get right on that.
Tom
--
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-- Bandwidth and Colocation Provided
On 01/25/2011 3:38 PM, Danny Nicholas wrote:
[snip]
Is there a good way to determine what version of SpanDSP I have
installed and whether the app_fax.so module is the same version?
[snip]
Try these two commands:
- whereis spandsp.so
- find /|grep spandsp.so
Those commands do point
On 01/26/2011 9:04 AM, jon pounder wrote:
On 01/26/2011 08:52 AM, Gilles wrote:
If you like open source what are you doing running windows ?
Getting anything to work properly there which does network
communications is a huge PITA since every user has their own firewall
and different settings
On 01/26/2011 1:49 PM, Kevin P. Fleming wrote:
snip
Steve did not write res_fax (which where SendFAX and ReceiveFAX come
from)
snip
I am personally a little confused here, because I have a ReceiveFAX
application when I unload the res_fax module and res_fax_digium module
and load the
On 01/26/2011 2:16 PM, Kevin P. Fleming wrote:
On 01/26/2011 01:12 PM, Tom Rymes wrote:
On 01/26/2011 1:49 PM, Kevin P. Fleming wrote:
snip
Am I correct to infer that using app_fax.so is no longer recommended and
that res_fax.so with res_fax_spandsp.so -OR- res_fax_digium.so is now
the way
OK, I generally use Hylafax+IAXModem for our faxing, but I have been
fiddling with FFA and SpanDSP for a while.
Is there a good way to determine what version of SpanDSP I have
installed and whether the app_fax.so module is the same version?
Many thanks,
Tom
--
On 01/21/2011 8:49 AM, Vitor Carlos Flausino wrote:
The system has 1 DAHDi card with 2 analog FXO ports (to pstn)
[snip]
However, it seams that when the call is received, the trunk does
not inform the DID
This is because FXO ports do not support DID. You need to route the call
based on
On 01/21/2011 8:59 AM, Steve Underwood wrote:
On 01/21/2011 08:37 PM, Tom Rymes wrote:
On Jan 20, 2011, at 8:52 PM, Steve Underwood wrote:
[snip]
Its easy to set up some t38modem channels and some iaxmodem channels for
receiving FAXes. Transmit is more problematic. With this split config
On Jan 19, 2011, at 11:08 PM, DSR wrote:
Is there anyway to play prerecorded agent intro-speech (like Hello, my name
is ) to outside caller when agent picks up?
I don't know of a way to do that, but I can say that, as a caller, it is highly
annoying. Your agents ought to be able to do
On 01/20/2011 10:58 AM, Jonas Kellens wrote:
[snip]
I have the following registrations :
register = 119909:pas...@sip.prov.org/52525252
mailto:119909:pas...@sip.prov.org/52525252
register = 119909:pas...@sip.prov.org/59595959
mailto:119909:pas...@sip.prov.org/59595959
[snip]
Problem :
On 01/19/2011 10:34 PM, Da Rock wrote:
WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to
non-existing call leg on other UA. SIP dialog
'481cf0543743e6bb7006991d409ed3bc@150.101.178.33:5060'. Giving up.
Have you tried disallowing re-invites?
--
On 01/20/2011 4:26 PM, Amit Nepal wrote:
I have an Audio code gateway between two asterisk servers. The audio
code has PRI connected for PSTN. I can send faxes and receive faxes in
ast 1.4 . Also I can send faxes for ast 1.6 to outside (PSTN) and
receive faxes. The only problem I am having is
On Jan 20, 2011, at 5:52 PM, Amit Nepal wrote:
On 1/20/2011 3:07 PM, Tom Rymes wrote:
On 01/20/2011 4:26 PM, Amit Nepal wrote:
I have an Audio code gateway between two asterisk servers. The audio
code has PRI connected for PSTN. I can send faxes and receive faxes in
ast 1.4 . Also I can
On Jan 19, 2011, at 3:18 PM, Jason Parker wrote:
On 01/19/2011 02:05 PM, Bryant Zimmerman wrote:
I am working on some fax tools for some of my users. I am reading the
https://wiki.asterisk.org docs for faxing.
Is see Application_SendFax and Application_SendeFax has one been
discondinued?
On Jan 19, 2011, at 10:06 AM, C F wrote:
On Sun, Jan 16, 2011 at 9:47 PM, James Miller paramedi...@gmail.com wrote:
When you get over 500 emails a day on your blackberry you have make a
decision on what is or is not worth reading at that moment.
Its not lazy at all its cutting through the
On Jan 19, 2011, at 5:06 AM, Vitor Carlos Flausino wrote:
In other words, which of the following is your situation:
1.) User dials 0X, asterisk sends 0X to the telco.
2.) User dials 0X, asterisk parses 0, strips it, and sends X
to the telco.
That might narrow it down.
On 01/18/2011 10:18 AM, Andrew Thomas wrote:
Why do I top post? Simple. I read every message in the thread - and if
there are 10 messages (for example) in that thread - then why should I
have to read them all over again on the last one?
OK, this is a stupid thread, nobody is going to be
On 01/18/2011 3:21 PM, Steve Totaro wrote:
If you are swapping out systems in really busy offices that rely on
faxing to keep the doors open, do a whole bunch of testing.
I have no experience with Digium's FFA, beyond installing it and
receiving a fax or two. So I can't really agree or
On 01/18/2011 3:20 PM, Vitor Carlos Flausino wrote:
== Spawn extension (DLPN_DialPlan1, 0924343424, 1) exited non-zero on
'SIP/6005-0002'
Vitor,
Can you please clarify whether the 0 should be received by Asterisk
and processed internally, or whether it should be passed to the DAHDI
On Jan 16, 2011, at 7:29 AM, bilal ghayyad wrote:
Dears;
I am looking for the card that does not need an electrical power, which one?
Is the PCI express doing this?
Regards
Bilal
Bilal,
The only telephony cards that require a power connector are those with FXS
ports for plugging in
On Jan 15, 2011, at 2:01 AM, Carlos Chavez wrote:
The problem is that Asterisk simply stops responding. No calls in or out
and you cannot even get to the CLI. The process seems to be running but there
is simple no activity. All I see in the log files is:
[Jan 14 16:30:46]
On Jan 15, 2011, at 9:29 AM, Don Kelly wrote:
That said, of course I want to follow this list's etiquette. I've posted a
couple times asking how I can interleave responses in Outlook or what other
approach can I take to make it practical to stop top-posting. Any
suggestions?
Don:
While we're at it, can someone please tell me whether I should be using
vi or emacs? ;-)
Many thanks,
Tom
PS: Bilal: You have asked a nearly unanswerable question. Some prefer
one, some prefer the other. Both cards are quality items. I can say that
I only have experience with Sangoma T1/E1
On 01/14/2011 4:19 PM, Bruce B wrote:
Also, to get the SIP very well as well, SIP uses both TCP/UDP 5060
right? and why are there recommendations of opening 5000-5082 UDP for
SIP along with 5060 TCP? Are there any niceties to that as well? maybe
video transmission stuff?
More likely, it's
On Jan 14, 2011, at 5:24 PM, Bruce B wrote:
So, simply pressing Reply and typing in the first line (using gmail webmail
without any clients) is a sin here? How is that top posting??? probably your
clients reading that way?
It may be a sin here, but it is certainly impolite many places,
On Jan 14, 2011, at 6:45 PM, Bruce B wrote:
You really want to read the LONG LONG signature from some people before you
read the actual latest message? I don't know about thatI guess it's a
preference.
Suffice it to say, Bruce, this subject has been hashed over thousands, nay,
hundreds
On Jan 14, 2011, at 7:12 PM, Bruce B wrote:
Thanks. That is in both TCP and UDP for SIP right? or simply UDP would do it
as well? I am talking strictly in case of Asterisk.
Asterisk 1.6 and newer support SIP over TCP. Older versions were UDP only, IIRC.
Tom
--
On Jan 14, 2011, at 8:52 PM, Don Kelly wrote:
I have nothing to add to the nascent flame war that I thought we had so
narrowly avoided when I sent my last message. However:
What did you mean, Andrew, about Don's multiple
signatures which I think he will review?
--Don
[snip]
Andrew meant
On 01/13/2011 11:25 AM, Danny Nicholas wrote:
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Thursday, January 13, 2011 10:19 AM
*To:*
On 01/13/2011 2:07 PM, Tom Rymes wrote:
That will require additions to your login/logout context that write
entries to the log each and every time a user logs in/out. You can then
report on that data.
While there's a thread going on about this topic, and while I've written
the above comment
On Jan 9, 2011, at 8:27 AM, William Stillwell wrote:
Anybody notice log delays in this list, and very small amount of traffic?
I have noticed multiple hour delays between sending messages and seeing them
back.
Tom
--
_
--
On Jan 6, 2011, at 8:08 PM, Joel Maslak wrote:
On Thu, Jan 6, 2011 at 7:06 AM, Jim Dickenson dicken...@cfmc.com wrote:
Are there reasons to prefer the use of PRI over SIP or SIP over PRI?
[snip]
I run the PBX for my organization which has about 160 extensions. I
wouldn't even think of
On Jan 6, 2011, at 8:56 AM, Kevin P. Fleming wrote:
On 01/05/2011 08:12 PM, Thomas Rymes wrote:
OK, after my last message about fax detection, I feel a bit better informed
and able to press forward. I started looking into this because I was getting
lots of false positive fax detection
On Jan 6, 2011, at 10:05 AM, Andy Graybeal wrote:
On 01/05/2011 01:51 PM, Tom Rymes wrote:
On 01/05/2011 7:50 AM, Andy Graybeal wrote:
We've got two noisy kitchens that need to talk back and forth.
Andy,
Why, exactly, are you trying to combine an inter-kitchen intercom and
your phone
On 01/05/2011 7:50 AM, Andy Graybeal wrote:
We've got two noisy kitchens that need to talk back and forth.
Andy,
Why, exactly, are you trying to combine an inter-kitchen intercom and
your phone system? Might it make more sense to have a non-phone-based
intercom system, plus a phone for
On 01/04/2011 8:55 AM, Kevin P. Fleming wrote:
On 01/03/2011 06:47 PM, Thomas Rymes wrote:
On Jan 3, 2011, at 3:22 PM, Kevin P. Fleming wrote:
On 01/03/2011 11:26 AM, Tom Rymes wrote:
[snip]
OK. Either way, though, the changes to echo cancellation are not
affected by the faxdetect setting
On 01/04/2011 8:52 AM, Andy Graybeal wrote:
Is it possible that I can run one cable to the phone, then run a cable
from the phone to a computer or another device and have those the phone
and computer or other device be on separate networks?
I'm sorry if this sounds newbish; I'm still learning.
On 01/04/2011 12:31 PM, Earl Terwilliger wrote:
Hi list,
I just installed Asterisk 1.4.38 (on an updated Centos 5.5 machine) and am
getting this error :
WARNING[6472]: res_musiconhold.c:856 moh_scan_files: getcwd() failed: No such
file or directory
[snip]
Have you installed mpg123 or some
According to https://issues.asterisk.org/view.php?id=16339 , the default
value for the dialdebounce parameter of the wctdm module has been
changed to 32 and is now user configurable.
I have two questions:
1.) Am I correct in presuming that, if the default of 32 does not work
for me, I would
On 01/03/2011 9:46 PM, Matt Watson wrote:
I don't imagine this would be too complicated - don't have any
experience with AsteriskNOW - but on a 'vanilla' linux distro it would
just be a matter of making sure dahdi is loading the correct drivers and
doing a couple of minor config file updates.
Hi folks,
I was hoping that someone might be able to help clarify some confusion I
have on DAHDI Fax detection after spending some time searching. My
understanding is this:
1.) Echo cancellation is automatically disabled upon recognition of a
CNG tone, regardless of the faxdetect setting.
On 01/03/2011 11:20 AM, Bruce B wrote:
Thanks a lot for that Sebastian. I will report back my findings when I
find the resolution on this.
I'm a bit late here, but I can say that Sangoma support has always been
extremely helpful when I call them. If you haven't already, definitely
give them
On Jun 20, 2007, at 5:04 PM, Troy Ayers wrote:
I would have been convinced if you had not top-posted! heh
Rob Schall wrote:
Tom,
I disagree with your argument for a number of reasons. Each of these
reasons should be more than enough to convince you I'm correct and
you
should do it my
On Jun 19, 2007, at 12:37 PM, Senad Jordanovic wrote:
Tom Rymes wrote:
[snip]
How many times does it have to be said? Don't feed the trolls!
Tom
Tom...Who in your opinion is a troll?
Senad
Well, technically, I was calling the original post a troll, not the
original poster. More
On Jun 16, 2007, at 4:37 PM, Tzafrir Cohen wrote:
On Sat, Jun 16, 2007 at 08:55:24PM +0100, Senad Jordanovic wrote:
Brett Crapser wrote:
On Saturday 16 June 2007 02:24 am, Senad Jordanovic wrote:
Paul Hales wrote:
GUI bad! CLI good!
PaulH
Really...?
So explain why every major PBX
On Jun 5, 2007, at 9:46 AM, Cosmin Prund wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Henry Cobb
Sent: Tuesday, June 05, 2007 4:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] NAT
On Jun 1, 2007, at 9:45 AM, Gordon Henderson wrote:
[snip]
Both these SIP - external PSTN provider connections register OK on
the * box, and outgoing calls placed over either connection works
perfectly. Outgoing callerId (set by the external provider) works
as expected. ) I have dialling
On May 24, 2007, at 3:28 PM, Doug Lytle wrote:
Paul Aviles wrote:
Hello guys,
I have been looking for a way to call a cell phone after someone
has left a
This can easily be done with database lookups and .call files
to accomplishing this? Most analog pbx's have this feature and I
am
On Thu, 24 May 2007, Jeremy Mann wrote:
Can an asterisk box equipped with a Digium T1 card handle
Integrated T1
circuits? I have a T1 with 768k data and the remaining channels
voice,
can the asterisk box do the Data routing + Voice processing?
I'm not certain, but I believe the
On May 5, 2007, at 12:06 PM, Rodrigo Mercado wrote:
Alguien tiene una TDM400P con modulo FXS usada a la venta ??,
obviamente a precio de tarjeta usada...
saludos,
Rodrigo Mercado S.
For anyone who is not a Spanish speaker, Rodrigo is looking for a
used TDM400P card with FXS modules.
On May 4, 2007, at 10:08 AM, Stephen Bosch wrote:
Tom Rymes wrote:
On May 3, 2007, at 12:20 PM, Stephen Bosch wrote:
Mats Karlsson wrote:
Take a look here:
http://www.voip.com.sg/voip_products/
voip_ip_phone_provisioning_tool.html
Ugh. This is a Win32 app, isn't it?
Wow,
The guy
On May 3, 2007, at 12:20 PM, Stephen Bosch wrote:
Mats Karlsson wrote:
Take a look here:
http://www.voip.com.sg/voip_products/
voip_ip_phone_provisioning_tool.html
Ugh. This is a Win32 app, isn't it?
Wow,
The guy makes a useful application and provides it to the community
for free and
On Feb 15, 2007, at 7:01 PM, Stefano Corsi wrote:
Hello everybody. First of all thanks to all the people giving their
opinion on the subject I proposed: Trixbox vs. custom install.
You've all been very helpful.
[snip]
I also include a consideration from mine: I would happily use
On Feb 12, 2007, at 7:13 PM, Stephen Bosch wrote:
Lee Jenkins wrote:
Stefano Corsi wrote:
[snip]
The nice things about GUI's in my opinion is that routine chores
such as
setting up extensions, dialing extensions, hunt groups, etc. are less
likely to contain scripting bugs or typos. The
On Feb 13, 2007, at 11:53 AM, Tzafrir Cohen wrote:
On Tue, Feb 13, 2007 at 10:23:17AM -0500, Tom Rymes wrote:
[snip]
Not to start a flame-war, but I completely disagree. Troubleshooting
a GUI is much easier, given that you don't have to scout for typos,
transposed numbers, etc throughout
On Jan 15, 2007, at 2:22 PM, [EMAIL PROTECTED] wrote:
Hello all,
we're using asterisk 1.2.12.1 in an Inbound callcenter using the
queue application. If there are many calls in the queue, it
sometimes takes up to 30 Seconds before a call is distributed to an
agent.
For example there are
On Jan 12, 2007, at 10:07 AM, Robert Norton - SophMedia LLC wrote:
Hey Guys,
I apologize for my ignorance on this one.
I've got several 7960s running on Asterisk1.4 with 15 or separate
queues and am trying to figure out a way to identify to the 7960s,
what queue the incoming call is on?
On Dec 7, 2006, at 4:14 AM, Jon Farmer wrote:
I decided to write my own simple voicemail application via AGI and
store all voicemails in MySQL. The nice thing was the user can
retrieve via phone (local and remote), via email attachment and
also via web download.
You can listen to old and
On Nov 30, 2006, at 8:55 PM, Brad Templeton wrote:
On Thu, Nov 30, 2006 at 02:50:21PM -0500, Tom Rymes wrote:
for example: In your example above where they can't figure out how to
transfer, why don't you edit features.conf and define the transfer
key as # or something. Then, when they have
On Nov 29, 2006, at 11:40 PM, Lacy Moore - Aspendora wrote:
[snip]
I went from a Lucent Merlin Legend system to Asterisk. For me,
it's a tradeoff for features. To my users, it was a step
backward. I also upgraded an office from a Partner system to
Asterisk. To the users, it is a huge
On Nov 20, 2006, at 2:21 PM, Don Pobanz wrote:
Eric ManxPower Wieling
No, you cannot change the volume of ONLY the dialtone
on a Zap interface.
I was afraid of that.
The most common problem with the first digit being
missed by the telco is that Asterisk is trying to
dial too soon after it
I have always been happy with Snapgear units. Most also have hardware based Encryption acceleration for IPSec and PPTP VPNs. As for setting up with SIP, use a VPN, or call their tech support line and they'll help you. Each unit comes with support for setup built-in last I
.
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Tom Rymes
Cascade Link Systems
or DNIS on Analog/TDM lines.
Tom
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414
Intelligent technology solutions for small businesses.
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Asterisk-Users
On Dec 13, 2005, at 8:25 AM, Michael George wrote:
On Fri, Dec 09, 2005 at 03:23:31PM -0500, Tom Rymes wrote:
On 12/8/05, chawki hammoud [EMAIL PROTECTED] wrote:
Hi:
i added these two lines to my general context ,but
nothing happened the same result the sound came in one
way for 3 seconds
and localnet entries in sip.conf,
You need to add a nat=yes entry
Tom
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414
Intelligent technology solutions for small businesses.
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waiting? That's how we handle this problem
with Cisco 79XX.
Tom
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414
Intelligent technology solutions for small businesses.
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, is that the
switch is just one more item for which you have to provide a backup
power source if you want your phones to work when the lights go out.
Those would be my recommendations, in order of preference.
Tom
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603
it helpful to go to asteriskathome.sourceforge.net and
spend some time perusing the handbook.
Tom
PS: If you haven't changed the passwords on your system, do it now!!
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414
Intelligent technology solutions
in the right direction.
Tom
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414
Intelligent technology solutions for small businesses.
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Asterisk-Users
).
DOH!
Don't know why I didn't try those, because I did see them there.
What's weird is that if I remove the entry for *99, I can now dial
*99 and it works. However, there is still a definition for *98 in
there as well, and that has worked all along.
Weird.
Tom
Tom Rymes
problem.
Tom
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414
Intelligent technology solutions for small businesses.
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of like Comedian
mail's externnotify parameter?
Tom
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414
Intelligent technology solutions for small businesses.
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To UNSUBSCRIBE
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To UNSUBSCRIBE or update
and there, that shouldn't be a big deal. If you
are talking about an office where lots of faxing is done, the lack of
reliability will likely be noticeable.
Tom
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414
Intelligent technology solutions for small
On Nov 26, 2005, at 4:35 AM, ram wrote:Hi thanks, but later i posted that, [EMAIL PROTECTED] need new server to start with as i have shortage of servers to go with that kind of setup so i have downloaded the Ast Source not [EMAIL PROTECTED] and compiled existing server setup, rather a new
with dyndns, and you need
to change the externip= option. Anyhow, if your address changes and
you do not update the dyndns service, the SIP client will be looking
for the wrong IP (the old IP.).
HTH,
Tom
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375
lose packets. YMMV, though, so try it out and see how it works for you.
Tom
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414
Intelligent technology solutions for small businesses.
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by O'Reilly http://www.oreilly.com/catalog/asterisk/index.html
Come to think of it, I have an extra O'Reilly official version of the
book that I will sell for $30 shipped. (Never used, I already have
another copy...)
Tom
Tom Rymes
Cascade Link Systems
then copy
the file to /var/spool/asterisk/outgoing and the call is executed as
defined.
Tom
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414
Intelligent technology solutions for small businesses
On Nov 25, 2005, at 7:00 PM, Manny A. Wise wrote:
-Original Message-
From: Tom Rymes [mailto:[EMAIL PROTECTED]
[snip]
On Nov 23, 2005, at 11:04 AM, Manny A. Wise wrote:
[snip]
Well, as the user stated on the original message, the asterisk
server is behind a NAT and the client is also
, for softphone use, I do indeed use IAX via
LoudHush for the mac. (Great piece of software, BTW. No connection
here, just a happy user...)
Tom
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414
Intelligent technology solutions for small businesses
. (Though I guess the DMZ setup would
have taken care of that...)
Tom
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414
Intelligent technology solutions for small businesses.
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BE
CAREFUL) and installs linux, Asterisk, AMP, etc.
You won't get your hands as dirty, and maybe not learn as much right
out of the chute, but it will be much closer to Just works than
installing Asterisk alone. (And your wife is likely to appreciate
this...)
Tom
Tom
On Nov 18, 2005, at 11:08 PM, Anton Krall wrote: I know you can send faxes using a hylafax client for windows and sending thru hylafax and iaxmodem out from asterisk but I was wondering, how do you receive faxes? can you received them as tiff and then convert to pdf and send via email or
for this, but this seemed simpler to me at the time I started. If it
can't be done using sh, then I'll start over in perl, but if there's
a way to make this work, it's the only thing standing between me and
a script that works exactly as I want.
Tom Rymes
Cascade Link Systems
to not ready/ready.
Wrapup is handled as a fixed time in the the queue setup for
Asterisk. You define wrapuptime=xx as a number of seconds. Asterisk
then waits that long before considering the agent to be available.
Tom
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
numbers?
Tom
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414
Intelligent technology solutions for small businesses.
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, though,
assuming that you get a way to interface it with your system:
Digium X100P
Digium TDM400P w/FXO Port
Digium TDM2400P w/FXO Port
ATA with FXO Port (Like Sipura SPA-3000)
Tom
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414
Intelligent technology
line to
asterisk, and a way for you to connect an analog phone to asterisk as
an extension.
It would be a good idea for you to spend some time on google, voip-
info.org, asterisk.org, asteriskdocs.org, etc. searching for
information.
Tom
Tom Rymes
Cascade Link
that mean it'll work... I just
looked on the Asterisk-Users message list and for some reason Tom
Rymes messages.
In response to Tom: I'm sure it's not an Digium-anything. It's a
cheapo Office Depot replacement for when my original was struck by
lightning. As I had said, I'm really not wanting
On Nov 14, 2005, at 2:50 AM, Dinesh wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom
Rymes
Sent: Monday, November 14, 2005 1:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Anybody tried
hiring a consultant to
help you out.
Tom
--
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414
Technology solutions for small and medium sized businesses.
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