As others have said, this is certainly possible. Our old NEC phone
system had us in the same boat. It triggered voicemail by ringing
the VM extension(s) and sending a DTMF burst of the extension to
record VM for within 1.5 seconds. In our case, when any call came it
in went to the
I use the following script to perform compression and normalization on
e-mailed voicemails. I put the script in as /usr/local/bin/sox and
pre-pend /usr/local/bin to the PATH before asterisk runs in the
startup script.
The values for the compressor are not scientific, I monkeyed with them
On Jun 18, 2009, at 2:57 PM, Philipp Kempgen wrote:
I think I would prefer this method, but I can't find where to set
asterisk to listen to the multicast address nor where to program the
notify reply
I have already told you that Asterisk is not involved in the process
of configuring the
but it is Digium's 2-port digital interface card
with HW echo canceler, using the wct4xxp driver.
Daniel
On Wed, Apr 29, 2009 at 17:29, Daniel Hazelbaker
dan...@highdesertchurch.com wrote:
Okay, I can't find what might be causing this. Here is what I got:
Asterisk server hooked up to a digital T1 line
Okay, I can't find what might be causing this. Here is what I got:
Asterisk server hooked up to a digital T1 line (full 24-channel) via a
Digium card.
Verizon has turned on caller ID on the first line (I can guarantee it
is on as I can hear the FSK tones on this line but not the others).
On Apr 27, 2009, at 10:29 AM, Danny Nicholas wrote:
Greetings all,
This is a “just-for-fun” question. I was
reading the support forum and a fellow there wanted Read() to stop
on * instead of #. I thought that changing app_read.c would resolve
this
current
if
On Mar 16, 2009, at 3:53 PM, SIP wrote:
David Ruggles wrote:
I was looking at the aastra 9133i, however I was informed that this
phone is
no longer supported. What are good phones around the $100 - $125
price
point? (Need POE)
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network
Is there a way to force a channel to continue in the dialplan after
the remote end hangs up?
Specifically, I am trying to play around with setting up a fax
server. I can receive the fax, but sometimes the sending fax hangs up
before my System command for printing can run and the fax never
On Feb 27, 2009, at 1:35 PM, Anthony Messina wrote:
On Friday 27 February 2009 14:03:19 Doug Lytle wrote:
Daniel Hazelbaker wrote:
Specifically, I am trying to play around with setting up a fax
server. I can receive the fax, but sometimes the sending fax
hangs up
If your looking
On Jan 16, 2009, at 7:52 AM, Julian Lyndon-Smith wrote:
Can anyone who has used both comment on the pros and cons ? Need to
buy
about 30 of these, for a small company with limited IT support.
We recently deployed 85 phones to our office. We tested the
Grandstream GXP2000, GXP2020,
I use the GXW-4008 and have never had any problems with it. Right now
it runs 3 analog phones, but we were using it to link our old NEC
phone system to the new Asterisk system, so it was used quite a bit
and never once had an issue.
Daniel
On Dec 24, 2008, at 5:30 AM, Hector Quiroz wrote:
We chose to use a mySQL database to store the holiday information.
When a call is answered we query the database to see if there is a
holiday greeting recorded, if so we play the indicated greeting,
otherwise play the default menu greeting. (We do our dialplans in AEL)
context
On Dec 2, 2008, at 7:01 AM, Grey Man wrote:
On Mon, Dec 1, 2008 at 3:26 PM, Steve Murphy [EMAIL PROTECTED] wrote:
Everyone--
I've just made some major changes to the CDRfix2.rfc.txt
file in http://svn.digium.com/svn/asterisk/team/murf/RFCs
to accommodate the Leg approach instead of a
On Dec 1, 2008, at 9:07 AM, JD wrote:
Steve Murphy wrote:
Freddi--
Very interesting. Brian Degenhardt had some code we just gave some
thought
to, wherein we determine if the last channel involved in a linkedID
set
has been closed. If so, then the entire set is finished. We can use
It will auto-complete if you hit tab, just like the shell. But I
would recommend against it. I can't really think of a good reason to
do it. 'sip show peer 268' I can remember to see that status of
extension 268 when somebody calls and says I can't dial 268.
Whereas 'sip show peer
On Nov 20, 2008, at 9:02 AM, Olivier wrote:
2008/11/20 Daniel Hazelbaker [EMAIL PROTECTED]
Any reason you want to use the MAC address? If it is just for easy
provisioning, I just put a MAC address field in the realtime SIP table
and use a php script to take the phone's MAC address and feed
I am suddenly getting a bunch of OLD (as in 3-9 months old) e-mails
from mantis saying things like a note has been added to an issue etc.,
and yet the issue has not been touched in months and the new note it
is referring to is also months old. Consequently, I never received
these e-mails
On Oct 28, 2008, at 5:13 PM, Kev Szaszvari wrote:
Hi there
Can anyone reccomend any voip phones( Cisco, Polycom, SNOM ) that have
* Central Management for all the phones (We dont mind if we have to
buy the software to manage them)
* Programable shortcut buttons, So i can program in on
On Oct 29, 2008, at 8:21 AM, Alex Balashov wrote:
In my experience most of the serious QoS issues arise in relation to
the
Internet pipe (if the provider is IP, and outside the network), not
the
LAN. Of course, LANs can be heavily contended, but are not in most
organisations,
On Oct 29, 2008, at 10:10 AM, Darrick Hartman wrote:
David Gibbons wrote:
Two separate networks? Did I miss something? I feel like I'm taking
crazy pills! Two separate physical networks means twice the hassle,
twice the maintenance, twice the cost, twice the headache. Not to
mention the fact
On Oct 24, 2008, at 9:49 AM, Wilton Helm wrote:
I've been following this thread and trying to sort out what is
wanted, what is available, and why. Comments to the following would
be appreciated and might be useful to others.
1. Why would anyone originate a FAX via VoIP? If it has to go
Might be a stretch, but does the Asterisk log show that the call was
answered? I had this problem when interfacing * with an NEC system to
do call parking pickup. The NEC would never give a dialtone (nor did
it give answer supervision) so * never knew the call got picked up so
audio only
On Oct 10, 2008, at 1:00 PM, Brent Davidson wrote:
Doug Lytle wrote:
I don't remember where I got it (Might have been the bug tracker)
that
works fine under the current 1.4.x. I had to do a minor change to
get
it to apply.
Copy into Asterisk source directory
patch -p0 *.patch
rm
On Oct 9, 2008, at 2:59 PM, Brent Davidson wrote:
I've got a situation where I need to use a transfer to the parking lot
as hold, but am not going to use BLF indicators on the phone to pick
up
the parked calls so I need to hear the 3-digit extension after the
transfer. I'm using Snom 300
On Oct 7, 2008, at 4:19 AM, Chris Bagnall wrote:
I recently purchased a few SRW208P switches. They work fine. If you
run Windows. Granted a lot of people run windows instead of Mac or
Linux, but be aware (to those looking) that the SRW line of switches
REQUIRE Internet Explorer on Windows.
On Oct 6, 2008, at 4:31 PM, Andrew Joakimsen wrote:
As for the larger switches I've used Linksys SRW224P. I have a few
running for a few years without issues. They have GB uplink but the
individual ports are 100M.
I recently purchased a few SRW208P switches. They work fine. If you
run
On Oct 2, 2008, at 9:10 AM, Jeff Peeler wrote:
- Tzafrir Cohen [EMAIL PROTECTED] wrote:
Yes, the new changes will be in 1.4.22. I continually have to
remind myself that users aren't running the most up to date code.
Once 1.4.22 comes out I will report if I am still having those
On Oct 1, 2008, at 11:39 AM, Jeff Peeler wrote:
Nope, that's the best you can do without restarting Asterisk. Is
requiring two restarts reproducible? I'd really like to see console
output with verbosity and debug set to 4 on chan_dahdi, preferably
while only using zap channels.
For
If I understand you, then yes you can. I do this now. All our telco
lines come through our analog NEC phone switch and then through FXO/
FXS ports to my Asterisk. Asterisk handles voicemail and the menu
system so when somebody dials 6 to get my extension the asterisk
does the following:
On Jul 11, 2008, at 12:58 PM, Daniel Hazelbaker wrote:
I may have figured out the problem this morning, but I won't be able
to test for a few days (again, aggravating that the only T1 line I
have to test with is the live one). I noticed this morning while
telneted into the Adtran that when I
On Jul 10, 2008, at 7:54 PM, Edwin Quijada wrote:
Hi! I am a newbie using Asterisk. I am developing an IVR using perl
from AGI and Cepstral as voices
The AGI is this
[snip]
My problem is that i cant hear anything when play the file sound
using $AGI-stream_file($filename);
I put
On Jul 11, 2008, at 10:08 AM, Douglas Garstang wrote:
I want to track call duration while the call is in progress.
To accomplish what? Are you wanting to beep the channel every 10
seconds? Are you wanting to play a you have 60 seconds left message
when they approach some quota? Are you
On Jul 11, 2008, at 12:09 PM, Jay R. Ashworth wrote:
On Tue, Jul 08, 2008 at 11:13:02AM -0700, Daniel Hazelbaker wrote:
D-Marc that terminates the 25-pair analog line coming in (this does
not just contain our lines as I can tap into other peoples lines and
hear there conversations, love
On Jul 11, 2008, at 1:31 PM, Edwin Quijada wrote:
vm-debian#file tts-hello
example.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM,
16 bit, mono 8000 Hz
Other than the filename being wrong which I would assume is the result
of a copy and paste from the original e-mail, that
out, but I don't think they will do anything other than say yep it
works on our side, fix your own equipment.
Daniel
On Jul 8, 2008, at 1:11 PM, Daniel Hazelbaker wrote:
Just an update for the information I got from Verizon:
It is a true T1, not a PRI for sure. b8zs and esf signalling
kinds of errors and completely
stopped working.
Date: Mon, 07 Jul 2008 16:55:27 -0400
From: Doug Lytle [EMAIL PROTECTED]
Daniel Hazelbaker wrote:
We are in the process of preparing to move our Asterisk server to a
Digital T1 interface card instead of a analog card (via an Adtran
which
On Jul 8, 2008, at 10:45 AM, Jay R. Ashworth wrote:
The Flex-grows I've seen were indeed T1, ESF as I recall the lights on
the Adit 600 they terminated them into.
Daniel: did Verizontal supply you with a shelf? Or just the
smartjack?
Uhhh... :) I have in my server room these things:
will report back afterwords if I had
success.
Daniel
On Jul 8, 2008, at 11:13 AM, Daniel Hazelbaker wrote:
On Jul 8, 2008, at 10:45 AM, Jay R. Ashworth wrote:
The Flex-grows I've seen were indeed T1, ESF as I recall the lights
on
the Adit 600 they terminated them into.
Daniel: did Verizontal
We are in the process of preparing to move our Asterisk server to a
Digital T1 interface card instead of a analog card (via an Adtran
which is now connected to the T1). I did a preliminary test the other
day and hooked the T1 line up to the T1 card, bypassing the Adtran.
This worked
to get elevated privileges. For me it is not a concern as the machine
is used only for Asterisk and only accessed by our IT department.
Daniel
CP
Daniel Hazelbaker wrote:
On Apr 1, 2008, at 5:22 PM, [EMAIL PROTECTED]
wrote:
Can the volume of the recorded voice mail message be changed? If
so
On Jun 30, 2008, at 1:04 PM, [EMAIL PROTECTED]
wrote:
But to get asterisk to run a different/fake sox, just install
whatever
you want to run as /usr/local/bin/sox and then edit your
safe_asterisk
script as I mentioned below.
I think this is a bad approach. It's going to be a big gotcha
On Apr 1, 2008, at 5:22 PM, [EMAIL PROTECTED]
wrote:
Can the volume of the recorded voice mail message be changed? If
so, what I am doing wrong? Any input would be greatly appreciated.
Thanks.
I had a similar problem in our setup where we e-mail the recorded
messages to e-mail
this? If not,
does anybody have a better suggestion for me? I'd rather not use a
regular digit as the begin code.
Daniel Hazelbaker
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(352) 392-0171 Ext. 221
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Daniel Hazelbaker
Sent: Wednesday, December 19, 2007 2:21 PM
To: Asterisk Users
. Obviously for other phones I would have to
come up with something else like the above, but with the grandstreams
it seems to work great.
Daniel
On Dec 19, 2007, at 11:48 AM, Daniel Hazelbaker wrote:
(Hope you don't mind me replying to the list)
Okay, time for me to feel stupid. Yes I
is that changed I make to the realtime database
don't get picked up immediately. Not sure what the cache timeout is
but I am able to flush it manually so for the moment I don't care. :)
Thanks,
Daniel
On Wed, 2007-11-28 at 16:56 -0800, Daniel Hazelbaker wrote:
I am trying to get the presence/hints/BLF
I am trying to get the presence/hints/BLF working along with Realtime
SIP but I never get any busy notification. core show hints always
shows the realtime sip user as idle. I have tried setting call-limit
to various values, including 1 but nothing seems to help. I have
tried limitonpeers
? :)
Daniel Hazelbaker
Information Technology Director
High Desert Church
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We have a Asterisk box acting as a voicemail system and greeting/
call director for our phone system (NEC system). The problem we are
having is that randomly (though most especially with cell phones)
asterisk thinks it is getting a double digit. For example, somebody
will enter
support they cross-shipped me a new card and the
problem (and that message) went away.
Daniel Hazelbaker
High Desert Church
On Jun 22, 2007, at 1:22 PM, Alex Mcdowell wrote:
HI I have two servers both of which get this message on one of the
lines.
Ring/Off-hook in strange state 6. The one
Have you quit and relaunched Asterisk? (not a reload, but a full quit
process and restart) I know in the past when I have a process
already listening to 0.0.0.0 it will not always pick up a newly added
NIC alias address without re-binding.
Daniel
On Apr 11, 2006, at 12:21 PM, Michael
Does anybody know how big a presence Asterisk and/or Digium will make
at Networld Interop this year? I have a part-time guy that is
building an Asterisk system for us (in a proof of concept fashion
before we do a full switch to it) that I would like to take, but I
don't want to waste his
contributed to this
discussion.
Regards,
Daniel Hazelbaker
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What I read on snom's website was the _currently_ only one sidecar
can be hooked up at a time. It sounds like they are working on
getting multiple sidecars chained together but have not got all of
the bugs worked out. I am kind of in the same boat. Our current
system offers 60 buttons
For those of us that only need a small handful of these receptionist
phones (for me it is 2), it should not be nearly as much of a
problem, correct? For example I only need 2 phones with 60 (well, I
can get 54 atm, but would like to expand even more). Assuming
everybody picked up their
like such a common thing.
Daniel Hazelbaker
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We may end up using a software solution, but there are two main
issues with a software solution (for us at least):
1) For us in particular, our receptionists have ALWAYS (for the past
15 years at least) used a physical switchboard style for routing
and seeing availability. From past
wrote:
On 3/27/06, Daniel Hazelbaker [EMAIL PROTECTED] wrote:
I have seen that the polycom setup (601+sidecar) works but only
for up to 7 phones
From what I've seen, each sidecar supports up to 14 additional
stations. Three of those along with the 5 buttons on the 601 comes up
to 47 on my
Hmm, which phone from Snom are you using for this? I've looked
around their website and I can only find 3 VoIP phones, the 300, 320
and 360. The 360 by the looks of it only has 12 buttons you can
assign to different extensions; am I missing something or is that the
phone and you just do
Drat, because the 3Com phones looked pretty good for the price. :)
Is there somewhere that has a compatibility list for Asterisk with
all the phones that are known to work/not work with Asterisk; since
apparently VoIP phone companies incorrectly state that they support
the SIP protocol (I
3101 (model with speakerphone)
3Com 3102 Business Phone
3Com 3103 Manager Phone
3Com 3105 Attendant Console (if these don't work, can somebody
recommend another receptionist alternative?)
Daniel Hazelbaker
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