*bump*
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Mike Hammett
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 20, 2008 4:52 PM
Subject: [asterisk-users] Coppercom and Asterisk
My provider has
It is, however, heavily trafficked and easy for someone to miss an email.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Benny Amorsen" <[EMAIL PROTECTED]>
To:
Sent: Monday, February 25, 2008 3:44 AM
Subjec
I thought it was odd, but I've had other devices work properly with that
information.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Chris Bagnall" <[EMAIL PROTECTED]>
To: "'Asterisk User
PROTECTED]
CSeq: 104 REGISTER
Expires: 120
Min-Expires: 900
Content-Length: 0
<->
--- (9 headers 0 lines) ---
-- Got SIP response 423 "Interval Too Brief" back from 63.164.210.14
Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER
--
*bump*
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Mike Hammett" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, February 21, 2008 11:55 AM
Su
I couldn't figure it out on my own. I tried to purchase a Smartnet for the
phone, but the original 7960 is not supported.
Is it technically possible and if so, what would it cost me to have someone
remote into my network and upgrade my SCCP 7960 to the latest SIP firmware?
--
I was doing it because of the volume on the server. It is very easy to miss
a message or 10 or 100 on a list of this traffic.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Tilghman Lesher" <[EMAIL PROTECTED]&g
That I am. I'll contact you off list.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Sigma Networks" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: T
or two
times I've had any issue, he has been quick to respond and took care of me.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Sigma Networks" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List -
message number 5. Can I just delete the 2 files and expect that
asterisk will renumber them? Or do I need to?
Also, is the answer the same when I migrate to storing voicemails in a
database?
Thanks in advance.
Mike
Unfortunately, I'm using a version of asterisk that is old enough to not
benefit from this...
Mike.
On Monday, October 9, 2023 3:15:45 PM EDT Michael Bradeen wrote:
> Hi Mike,
>
> New AMI actions were recently added to app_voicemail to let you remotely
> manipulate a
John, that is some serious script-fu! I does exactly what I was going to do
in perl. However, my initial testing indicates that asterisk will renumber
voicemail boxes to eliminate holes. But I'm still testing.
Thanks again,
Mike.
On Tuesday, October 10, 2023 11:47:35 AM EDT John Har
respond to
ping, so it's not completely dead. I've had the same symptoms with
SPA303's sitting behind Ubee modems.
So, is there some configuration setting on the SPA that I can set to make
this device mor
#x27;s
not an option in this particular case, though.
Mike.
On Thu, Feb 6, 2014 at 11:27 AM, Leandro Dardini wrote:
> How long is the registration timeout? If the device is behind a
> router/firewall, then you need to set a registration timeout lower than the
> state table "life&
Unfortunately, we plug straight into the Ubee and the ISP will not support
any other modem.
GRRr..
Mike.
On Thu, Feb 6, 2014 at 12:34 PM, David Wessell wrote:
> Is there another router in the mix? Put the cable modem in bridge mode and
> attAch a real router.
&g
Based on what we're hearing, we've decided to replace the SPA112. Thank
you for your input.
Mike.
On Thu, Feb 6, 2014 at 4:39 PM, Andres wrote:
> On 2/6/14, 11:18 AM, Mike Diehl wrote:
>
> Hi all,
>
> I have an SPA112 that in sitting behind a Ubee cable modem
e call. My logs only indicate
that the 'h' extension was called once, at 9:29:07
My question is, how can a call not get hung up when both parties hang up
the call? I know that sounds odd, but that's what I'm seeing.
Any ideas?
Mike.
--
___
h extension calls an agi scrip that logs a bunch of information about
the fax attempt. Works just fine when I receive a fax. But there is no
sign of it in the logs for the sending leg of the fax.
Is there something I need to do in order to get
the fax, but not
the final delivery.
Does that make more sense?
Mike.
On Wed, Feb 19, 2014 at 6:10 PM, Matthew Jordan wrote:
> On Tue, Feb 18, 2014 at 2:13 PM, Steve Edwards
> wrote:
>> On Mon, 17 Feb 2014, Mike Diehl wrote:
>>
>>> Is there something I need to do in
user?
Any ideas?
Mike.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users ma
I'm sorry, I should have mentioned that he's doing a "phone-based"
transfer, not an "asterisk-based" transfer.
Mike.
On Mon, Feb 24, 2014 at 1:30 PM, Don Kelly wrote:
> Does he complete the call as a "supervised" transfer--waits for the called
>
is is causing our users to not get a positive acknowledgement when
they send the fax.
Is there anything we can do to mitigate this?
Mike.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Ast
Steve,
I BELIEVE the fax is complete because the fax image is a form that appears
to only be a single page.
But, since FFA isn't providing acknowledgement, the sending fax machine is
resending the document multiple times!
Mike.
On Mon, Mar 10, 2014 at 12:49 PM, Steve Underwood wrote:
&g
, then.
Thanks again.
Mike.
On Tue, Mar 11, 2014 at 12:27 AM, Steve Underwood wrote:
> Hi Mike,
>
> If the sending machine keeps trying it might be the call has been hung up
> by asterisk before its own acknowledgement message has finished being sent.
> There have been bugs like th
flexibility and
stability as I can get.
So, what are your recommendations?
Mike.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
trouble.
So, where else can/should I look?
Mike.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk
e T.38 for line 1.
Yes
1
Yes
caller or callee
This is what I'm using to turn cfwd off on line 1.
No
No
No
Yes
Yes
This is what I'm using to turn call waiting off on line 1.
No
However, these setting don't seem be be getting set on the device, even
after a reboot.
Any
Well, I went to an online xml validation site and found an error. After
correcting the error, my problem is gone!
Thank you.
Mike.
On Thu, Mar 27, 2014 at 2:56 PM, Noah Engelberth
wrote:
> To me, the settings you've sent look correct. However, one thing I've
> found with S
l to come from phone A. What can I do? I
really dread putting each phone into their own context and parameterizing
their ID...
Any ideas?
Mike.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to A
ding line supervision and
the PBX requires it.
Does anyone know how to fix this? I'd also like to fix it from a
provisioning file, if possible.
Thank you!
Mike.
--
_
-- Bandwidth and Colocation Provided by http://www.api-
0 Threshold APIC interrupts
MCE: 0 0 Machine check exceptions
MCP: 24 24 Machine check polls
ERR: 1
MIS: 0
Should I just give up on using the card in this server ?
Is there anything else I can try ?
What other information may be r
.040% 88.970% 89.148%
--- Results after 40 passes ---
Best: 89.559% -- Worst: 88.573% -- Average: 89.052215%
Cummulative Accuracy (not per pass): 89.052
Still experimenting.....
Best regards,
Mike
On Mon, 2014-05-12 at 17:23 -0500, Russ Meyerriecks wrote:
> On Mon, May 12, 2014 at 4:5
read up a bit more on the subject and look at possible power
saving issues on this machine.
Best regards,
Mike
On Tue, 2014-05-13 at 15:26 -0500, Russ Meyerriecks wrote:
> On Tue, May 13, 2014 at 7:28 AM, Mike Leddy wrote:
> But on examination the /etc/init.d/dahdi start wa
running out of ideas. Please help.
Thanks,
Mike
On Tue, 2014-05-13 at 17:56 -0300, Mike Leddy wrote:
> Thanks again Russ,
>
> Just a quick reply for now.
>
> No virtualization, but yes I am running a tickless kernel:
>
> #
> # Processor type and features
> #
> CO
rface so I can use it in recent servers but it uses an
older chipset and driver than I was using.
Thanks for the help,
Mike
On Wed, 2014-05-14 at 15:54 -0400, Eric Wieling wrote:
> Try the card in another machine with a different brand of motherboard. If it
> works you know it is
ample period: 10.013 s
10014, 9027 modprobe init_module (dahdi_dummy_hr_int)
I will test it on a live E1 soon.
Best regards,
Mike
On Wed, 2014-05-14 at 16:53 -0500, Russ Meyerriecks wrote:
> On Wed, May 14, 2014 at 3:41 PM, Mike Leddy wrote:
>
span 1
Not usable in production but getting a lot closer.
Is there anything else that can be done to improve this ?
Best regards,
Mike
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk
wc
28:1700363 0 0 0 0 0
0 0 0 0 0 0 IR-IO-APIC-fasteoi
wcte11xp
28:1701370 0 0 0 0 0
0 0 0 0 0
[4017] chan_dahdi.c: PRI got event: Alarm (4) on
D-channel of span 1
[May 15 17:36:25] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on
D-channel of span 1
Best regards,
Mike
On Thu, 2014-05-15 at 17:53 +0100, Gareth Blades wrote:
> On 15/05/14 16:28, Mike Leddy wrote:
> > Hi Russ,
>
period elapses. But they don't.
Any ideas?
Mike.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.ast
f my SPA112's are running 1.3.2(014). My SPA8000's are running 5.1.10.
--
Mike Diehl
Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505) 903-5701
--
_
-- Bandwidth and Colocation Provided by http://www.api-dig
On Tuesday, August 05, 2014 05:19:55 PM Steven Howes wrote:
On 5 Aug 2014, at 17:10, Mike Diehl wrote:
All of my SPA112's are running 1.3.2(014). My SPA8000's are running 5.1.10.
If you do firmware upgrade your 8000s, don’t go past 6.1.3 or it’ll go badly…
Freezing and requi
isk versions and their supported lifetimes,
> please see the following wiki page:
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
>
> Thank you for your continued support of Asterisk!
Is there any time frame for when FFA will be available for 13?
--
Mike Dieh
0/0.0.0.0
public
private
0
v3rwuser
MD5
11
DES
11
-08 1 1
1
auto
3600
0
1
0
1
0
0.0.0.0
1
1
0
1
1
1
1
1
0
1
0
1
1
0.0.0.0 0
80
0
86400
1
0
0
200
syslog.example.com
514
25
100
60
0
3
0
0
0
0
3
0
0
0
0
admin
cisco
that is stable. In
the mean time, I'm trying to run unixODBC 2.3.2.
What version SHOULD I use?
TIA,
--
Mike Diehl
Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505) 903-5701
--
_
-- Bandwidth and Colocatio
GI script looks, essentially, like:
$main::agi->exec("ConfBridge","1505xxx");
I've got a dummy /etc/asterisk/confbridge.conf file:
[general]
[default_bridge]
type=bridge
[default_user]
type=user
[default_bridge]
type=bridge
[1505xxx]
type=bridge
Any su
ax; check the manual that corresponds
>
> On Mar 23, 2016 11:38 PM, "Mike Diehl" wrote:
> > Hi all,
> >
> > I've got a new server up, but it's not staying up
> >
> > After a day or so, it segfaults with:
> >
> > [Mar 22 23:
le structure do I use?
Thanks in advance,
--
Mike Diehl
Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505) 903-5701
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join u
bridge to be created?
Thanks again,
Mike.
On Saturday, April 16, 2016 04:18:44 PM Bobby Hakimi wrote:
> You can't see them until someone joins the bridge, might be able to put in
> db using the asterisk live setup
>
> On Apr 16, 2016 1:36 PM, "Mike Diehl" wrote:
>
Hi all,
I've got a device that seems to become unreachable for about 2 minutes, every
hour. From what I can tell, it isn't due to network or server issues. Any
ideas?
TIA.
--
Mike Diehl
Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505
produce and I can't emulate easily.
Any recommendations would be very welcome.
--
Mike Diehl
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: htt
produce and I can't emulate easily.
Any recommendations would be very welcome.
--
Mike Diehl
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: htt
es; nothing resolved the issue.
Any suggestions?
--
Mike Diehl
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to
ia RT using
a local Mysql database.
We only use the native SIP channel driver at this time.
I honestly don't see any reason for this server to eat 100% of it's cpu, and
am hesitant to roll it out to production until I understand why it is.
Once again, any suggestions will be welcome.
Dear Saint Michael,
I will be grateful if you could introduce me to the Company that
offers the translation service.
I am really interested in google voice.
Sincerely,
Michael Codjoe
On 29 March 2017 at 17:00, wrote:
> Send asterisk-users mailing list submissions to
> asterisk-users@
on my part.
Hope this helps someone else.
Mike.
On Thursday, April 06, 2017 10:28:03 AM you wrote:
> On Thu, Apr 6, 2017 at 10:20 AM, Mike Diehl wrote:
> > I found it!
> >
> > I had customized the safe_asterisk script and managed to slip in a -c on
the asterisk command li
7-r1
I suspect that the odbc drivers are the problem. Is ther an alternative drive
that I should be using?
Failing that, any other ideas?
Thanks in advance.
--
Mike Diehl
--
_
-- Bandwidth and Colocation Provid
Thank you for your time. I've put my replies to your questions in-line, below.
On Wednesday, June 07, 2017 10:19:41 AM Antony Stone wrote:
> On Tuesday 06 June 2017 17:54:59 Mike Diehl wrote:
>
> > Hi all,
> >
> > I'm upgrading to Asterisk 13.14.0 x86
I could find out what syscall was being interrupted That MIGHT
tell me what was wrong, but this is all I get from strace.
Any ideas would be welcome.
Mike.
On Wednesday, June 07, 2017 04:34:10 PM Mike Diehl wrote:
> Thank you for your time. I've put my replies to your questions in-li
in.
I'm running Asterisk 13.14.0 built by root @ server on a x86_64 running Linux
on
2017-06-20 14:27:06 UTC
For odbc, I've got unixODBC 2.3.2-r2.
Are these the versions I should be using? If so, any recommendations as to how
to
troubleshoot this would be
Man, I was hoping it was something like that. I did read the release notes; I
must have missed that part.
This should solve the problem, so thanks again.
Mike
On July 20, 2017 1:09:08 PM EDT, Richard Mudgett wrote:
>On Thu, Jul 20, 2017 at 11:50 AM, mdiehl
>wrote:
>
>> I re
ike the idea of the IP geo location firewall rules coupled
with the "friendly scanner" filter, as provided by a few of you guys. It
was mentioned that this is a broad hammer, but I'm kinda looking for a
broad hammer! ;^)
Looks like I need to do some research, but I think I have what I need.
actually works between GS phones or not, but it
definitely doesn't to other stuff. They negotiate gsm fine but send
gsm data to the rtp port and the GS phone replies with icmp errors.
Non-gsm data is fine...
Mike
P.S. Asterisk <-> (say) X-Lite using gsm is fine...
--
Mike Jagdis
I am trying to get the handytone 286 to make a very simple call to * and
having problems. It registers with * just fine, but when I place a call
(to echo test, for example), the RTP stream seems to have problems
opening. Here is there error I get in *:
WARNING[98311]: File chan_sip.c, Line 464 (re
error.
> It took three days to track down the error to buggy network hardware.
>
> Same symptoms, too - phone registered, one way conversation was ok (had a
> test extension
> for music on hold)
>
> Fixed the hardware, phone works great.
>
> John
>
> - Origi
handytone,
handytone sends NO RTP packets to *, so maybe it does not get the clue.
On Sun, 2004-01-04 at 20:16, Mike Machado wrote:
> I am trying to get the handytone 286 to make a very simple call to * and
> having problems. It registers with * just fine, but when I place a call
> (to echo test
FYI there is a way to do 911 its called E-911 enhanced 911
the user has to set it up with the local emergency services
to it and you setup your pbx to xmit the data.
Here is the fcc rule about it
http://www.fcc.gov/911/enhanced/
___
Asterisk-Users m
On my handytone, if I did not enable STUN, the * box would send the RTP
data to my 192.168 address, even though I had nat=yes in sip.conf and
the SIP handshake happened with my public IP. It seemed * was not
properly translating (ignoring) the IP that was in the SDP into the IP
that the SIP handsha
)
<< Sorry, had to cut the dump due to size
restrictions, but is available upon request >>
-- Mike “Dexter” Church
iller feature(s) that you feel this
product should have?
--
Mike
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/ast
Hi can anyone help me with this
g++ -g -c -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN
-DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC
-Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT
-D_GNU_SOURCE -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES
-DP
he source again, but I get the same results.
I'd appreciate any pointers that you may be able to give me as I don't really
know where else to look.
Thanks,
Mike
-
[EMAIL PROTECTED] libpri]# make install
cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o pri.
Thanks to Dave and Matt for your help. I now have Asterisk running on my
Mandrake 9.1 box - time to learn how to configure it :)
Regards
Mike
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
26]: chan_sip.c:5548 handle_request: Registration
from 'Mike ' failed for '203.118.186.16'
I've tried looking at the www.automated.it setup information, along with the
information on fnords.org - this has gotten me this far, but I can't see for
the life of me what I
that playing a track in XMMS seemed to stop the
Asterisk error.
My SIP phone is located on my server at the office, and my Asterisk server is
at home - both behind NAT'd firewalls.
As you suggested, I have taken the sample config and will try and experiment a
little from here.
Regards
Do both the budgetone and the handytone use the same firmware?
On Fri, 2004-01-30 at 06:26, Stephen R. Besch wrote:
> Greg Boehnlein wrote:
>
> > On Thu, 29 Jan 2004, Michael Welter wrote:
> >
> >
> >>I have 1.0.4.45 (beta) on my tftp server. Try it at 66.250.23.58.
> >>
> >>Cheers,
> >>Micha
Bob, I have a question into mediatrix for this, but maybe you have
figured it out. I am trying to map a SIP user to a specific PSTN line. I
have my extensions.conf file as you show below, but on the 1204, it just
grabs whatever line is available, whereas I want extension 101 to always
be port1 on 1
> I'm coming at this thing from an Operational standpoint rather than a
development standpoint. Viewing your problem from that angle, I wonder how
well your
> network is performing. Could you have a cable problem that the Asterisk
server hasn't reported (Layer 1); or perhaps your * Server is conn
Any other thoughts?
Thanks!
Mike
- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, October 07, 2003 11:32 AM
Subject: Re: [Asterisk-Users] IAX and Jitter problem
> On Tue, 2003-10-07 at 11:14, [EMAIL PROTECTE
My understanding is that the ascend gear only speaks IPDC and not MGCP,
so not sure it would even work with asterisk.
On Thu, 2003-12-04 at 15:09, Steve Dolloff wrote:
> I would be seriously wary of putting a DS3's worth of voice traffic on a
> TNT. I don't believe they are rated to handle that m
Yeah, recompiling the kernel is a bit over my head, but I don't want to
install an older gcc, so I'll just have to await some hand-holding from the
people that put my kernel together (OpenVZ).
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Origin
vious answer to these questions. I'm
just
trying to get my feet wet with Asterisk for the first time.Thanks,Mike
Do you Yahoo!?
With a free 1 GB, there's more in store with Yahoo! Mail.___
--Bandwidth and Colocation provided by Easyne
I'm running a VPS and I need to pass the device
drivers from the host OS to the VPS. What files do I need to pass through
for ztdummy to work? I'm assuming they're in /dev/zap, but I'm not sure
which ones are needed.
Mike HammettIntelligent Computing
Solutionsh
7;t have to run multiple wires to the phones and makes for a neater look. I dislike the looks of the SNOM series, so that's out.Have the bugs been worked out with the latest firmware, or are these still phones to avoid. thanks,Mike
Do you Yahoo!?
With a free 1 GB, there's more
I just saw the S100-FX v2.0 on eBay. I was wondering if anyone has tried it out
and what their opinion of it was.
Mike HammettIntelligent Computing
Solutionshttp://www.ics-il.com
___
--Bandwidth and Colocation provided by Easynews.com
It does use the same kernel for everything. It's a specially modified
kernel for the VPS support. I guess the only way to see if ztdummy works in
the VPS is to try it.
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: &l
What about IAX - SIP or IAX - IAX?
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: <[EMAIL PROTECTED]>
To:
Sent: Saturday, January 28, 2006 5:43 AM
Subject: Asterisk-Users Digest, Vol 18, Issue 185
Send Asterisk-Users m
directory has the files channel, ctl, pseudo, and timer. Which
of those do I need to pass through? Are there more elsewhere?
The ztdummy is actually running on the host OS, which is not affected (IIRC)
by any virtualization controls, so the interrupts/second shouldn't be an
issue.
ings support it.
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: <[EMAIL PROTECTED]>
To:
Sent: Thursday, February 02, 2006 2:52 PM
Subject: Asterisk-Users Digest, Vol 19, Issue 19
Send Asterisk-Users mailing list submissi
helps,
Mike
- Original Message -
From: "Nora Lavelle" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, February 03, 2006 3:13 PM
Subject: RE: [Asterisk-Users] Calls fading in and out
Hi John - Good call. yes I&
for a limited number of channels.
Mike HammettIntelligent Computing
Solutionshttp://www.ics-il.com
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http
Can TDMoE be used for non-voice
applications?
Can another box be setup with TDMoE on the other
side to dump it back out via T-1?
How does this compare with an off-the-shelf TDM
over Ethernet or IP device?
Mike HammettIntelligent Computing
Solutionshttp://www.ics-il.com
Reason I ask is I may have a non-voice T-1 replacement project going on and
I'm investigating my various options. Costs may be about the same for
turn-key and DIY.
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: <[EMAIL P
Ethernet instead of IP is its only issue.
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: <[EMAIL PROTECTED]>
To:
Sent: Thursday, February 09, 2006 3:11 AM
Subject: Asterisk-Users Digest, Vol 19, Issue 59
Send Asterisk-Us
her brands because
I've never used them.
...snipped a lot ...
The Sangoma A101 works well too.
Mike Clark
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://l
two weeks since it
has been installed. We've got another one on the way for a second site.
Mike Clark
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Case
sensitivity? The CLI references Goodbye but your filename is goodbye.gsm.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam Lee
Sent: Friday, 10 February 2006
1:22 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Hi Gerard --
I found that I get the really loud buzzing sound in the handset earpiece
when I set echocancel=256 instead of echocancel=yes (the default = 128
taps).
It seemed to occur irrespective of the actual echo canceller chosen.
Mike.
-Original Message-
From: [EMAIL PROTECTED
Hi
That's a known problem with 1.2.2. Upgrade to 1.2.4.
Mike.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kamran Ahmad
Sent: Saturday, 11 February 2006 9:09 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] No Voice when canrei
,
Mike.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Lith
Sent: Sunday, 12 February 2006
7:51 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
TE411P Really Bad Echo
Sorry, that's correct -
so when experimenting wit
701 - 800 of 2253 matches
Mail list logo
