- Original Message -
Hi
Thank you for your support.
The server is actually compromised, I discovered that after making a
deep trace using the audit daemon and looking for the kill signal
(SIGKILL) that terminates asterisk.
I discovered that there is an executable with a random
- Original Message -
On 10/22/2014 03:55 PM, Tim Nelson wrote:
- Original Message -
Greetings-
Working with the T.38 gateway functionality that is sparsely
documented [1], I'm attempting to get the following functional:
Asterisk calling system - Asterisk system
- Original Message -
On 23/10/2014 3:55 AM, Tim Nelson wrote:
- Original Message -
Greetings-
Working with the T.38 gateway functionality that is sparsely
documented [1], I'm attempting to get the following functional:
Asterisk calling system - Asterisk system
- Original Message -
On 23/10/2014 10:07 PM, Larry Moore wrote:
On 22/10/2014 11:23 AM, Tim Nelson wrote:
Greetings-
Working with the T.38 gateway functionality that is sparsely
documented
[1], I'm attempting to get the following functional:
What type of endpoint
- Original Message -
On 22/10/2014 11:23 AM, Tim Nelson wrote:
Greetings-
Working with the T.38 gateway functionality that is sparsely
documented
[1], I'm attempting to get the following functional:
What type of endpoint are you using which is originating the call
- Original Message -
Greetings-
Working with the T.38 gateway functionality that is sparsely
documented [1], I'm attempting to get the following functional:
Asterisk calling system - Asterisk system in T.38 Gateway Mode (box
in question) - SIP Provider
The problem is:
-The
Greetings-
Working with the T.38 gateway functionality that is sparsely documented [1] ,
I'm attempting to get the following functional:
Asterisk calling system - Asterisk system in T.38 Gateway Mode (box in
question) - SIP Provider
The problem is:
-The provider is not initiating a
- Original Message -
Tim,
I THINK but I'm not sure that you can do this with the Polycom
multicast page function. Have you attempted this yet?
Thanks
david
Given the odd nature of multicast paging with Polycom, I was hoping to avoid
such a setup. My recollection is having this
Greetings-
As many of your are Polycom experienced, I was hoping some kind soul could
provide direction on a specific issue.
On a system running Asterisk 11.11.0 (and latest FreePBX), I'm finding an
instance where, using intercom/paging functionality of FreePBX, I need to
override an end
- Original Message -
On 13 Feb 2014, at 18:10, Tim Nelson tnel...@rockbochs.com wrote:
I recently experienced an odd situation. I have an Asterisk 11.5.0
system (Box A) with a SIP peering to another Asterisk 1.8.23.0
system (Box B). At some point, Box A started sending over 65Mbps
- Original Message -
SIP options message is due to check the peer registration is
keepalive. As per my understanding it might be because of network
flap may be wireshark trace can give you any clue.
Regards
Correct. I understand the role and function of the OPTIONS requests. The
Greetings-
I recently experienced an odd situation. I have an Asterisk 11.5.0 system (Box
A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At some
point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box A. I do
have qualify=yes for the peer on both sides, and
- Original Message -
http://camrivox.com/products/flexor-cti-salesforce/
We've used this for a few clients.
How were your experiences with it? I have a customer that will want this type
of integration in the near future, and would love to hear how installation,
operation, and
Greetings-
I have an odd scenario where I need to dial an extension (lets call it 555),
the system prompts for a list of voicemail boxes, then once complete, allows
the caller to leave a voicemail that is sent to all voicemail boxes previously
specified.
How would you do this? Obviously
- Original Message -
Is anyone aware of a way to replicate parts of the AstDB to another
Asterisk install?
For example, to export all CF entries on a FreePBX based system to
another system running FreePBX, I might do:
asterisk -rx 'database show' | grep CF
This gives me a list
Is anyone aware of a way to replicate parts of the AstDB to another Asterisk
install?
For example, to export all CF entries on a FreePBX based system to another
system running FreePBX, I might do:
asterisk -rx 'database show' | grep CF
This gives me a list of data, which I can rsync to
- Original Message -
No, my phones aren't getting a response from the server. I can't
even
get any output from the server if I do:
sip show peer name load
This command usually loads the peer from the db and shows me it's
configuration. In this case, I get nothing.
I do have
Greetings-
I'm running some USB DAHDI hardware on a system with a tickless kernel. The
audio quality is quite poor. Could the tickless kernel be to blame? If so, when
recompiling a kernel that is *not* tickless, is there a recommended KERNEL_HZ
value? IIRC, older kernels used to be 1000, but
- Original Message -
On Tuesday 28 May 2013, Tim Nelson wrote:
Greetings-
I've got a curious project that I could use some input on. I'd like
to use
Asterisk to record some audio channels via USB 'soundcard'. When
audio
passes through the soundcard, Asterisk should grab
- Original Message -
You are still being a bit evasive but should I understand that you
want
to run a headless machine with open microphones that records what
ever
it hears?
What do you want to do with each sound bite?
How long does the silence have to be before you close the
- Original Message -
I'll take a stab, since you said no GUI and also USB based mic.
Raspberry Pi project? I'm interested in this vein as well,
Nope, the RPi is not a component. BUT, a good suggestion for a small board.
I'll keep it in mind.
especially
after the recent post about
Greetings-
I've got a curious project that I could use some input on. I'd like to use
Asterisk to record some audio channels via USB 'soundcard'. When audio passes
through the soundcard, Asterisk should grab that audio channel (CONSOLE?), and
write it to a wav file. I'm perfectly competent
- Original Message -
What are you trying to accomplish?
What is the USB 'sound card' attached to?
Your description is too cryptic for someone to propose a solution.
The target use is to record mic level audio from various devices (could be an
omnidirectional room mike, phone
- Original Message -
Sorry for the blank message. Fingers pressed send while brain was
disenaged.
Would Audacity be a better choice?
http://wiki.audacityteam.org/wiki/Multichannel_Recording
It would absolutely be a better solution. However, the recording is to be
automated on a
First thing to *ALWAYS* check is if you have any Asterisk version specific
modules (Fax for Asterisk, G.729, etc). Ensure these are not loaded (noload in
modules.conf, or simply move them out of the asterisk modules dir).
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
I'm getting this error message on my Asterisk CLI, and in the logs, roughly
every 10-20 seconds:
[2012-12-18 19:19:51] ERROR[24485]: astobj2.c:115 INTERNAL_OBJ: user_data is
NULL
While it doesn't appear to be actually affecting anything, I'm curious to know
what the error represents, where
- Original Message -
Tim,
What version are you on? There is a specific upgrade path for pre
3.3.
Yes, that was the issue. I needed to upgrade to version 3.3 first, then upgrade
to latest 4.x was no problem. Thanks!
--Tim
--
* the new firmware placed there. So, is the Polycom firmware matrix
wrong about this phone/firmware compatibility, or am I missing something? The
bootrom has also been upgraded to the latest without any problems.
Thoughts? My head is getting sore from banging it on my desk... :/
Tim Nelson
Systems
- Original Message -
Switching to SIP is likely your best solution. IAX is buggy. Always
has been, and I'll bet always will be.
Alright, I'll bite on this one.
Can you give any specifics about IAX being buggy, other than throwing out
random claims? I understand it doesn't get the
- Original Message -
I wish to ask if there is way to keep IAX trunk connection up. I have
a small server on Xen VPS but notice that my IAX trunk drops after
some time.
I understand there is cron job to function as sip watchdog.
My asterisk is 11.0.1
You'll want to use
- Original Message -
Tim Nelson wrote:
Greetings-
Hola,
I'm running into an issue as follows, in simplified form:
A remote Asterisk box, when registered/peered via SIP to a central
server, and makes a call to that central server, is *sometimes*
authenticated and calls go
Greetings-
I'm running into an issue as follows, in simplified form:
A remote Asterisk box, when registered/peered via SIP to a central server, and
makes a call to that central server, is *sometimes* authenticated and calls go
through properly (via from-internal context), and *sometimes* is
- Original Message -
Has anyone seen issues with recent Sangoma T-1 cards and Sangoma
Analog cards on multiple different servers?
On T-1: we get NO traffic, no interrupts, and no increase in number
of packets and the PRI does not come up.
On Analog: The ports do NOT go red when
- Original Message -
Have a look at your /etc/asterisk/rtp.conf file. In it you specify
the UDP portrange your asterisk will use for RTP traffic. change the
rtpstart and rtpend to your needs and set them open in your FW. Do
not make the range too small each active call will normally
- Original Message -
No idea? ):
How about showing your dialplan, and the log or console output from when you
make the call? I have a hard time believing this number is special in any
way...
--Tim
--
_
--
Is there a way to have Asterisk respond appropriately when receiving a DTMF
Flash event via SIP? I'm finding some WiFi SIP phones, specifically the
Quickphones QA-342 want to handle transfers/3-way calls by sending a DTMF Flash
event instead of handling it properly like every other damn VoIP
- Original Message -
On Thu, 20 Sep 2012, Jerry Geis wrote:
Actually I restart asterisk every day at 2AM. So something happens
in a
24hour window.
On Thu, 20 Sep 2012, Jerry Geis wrote:
THanks, actually all of my modifcations were to the extensions.conf
file
itself. It
- Original Message -
Yeah, I noted that too, but besides that it seems like it is exactly
what I am looking for. I am especially confused that there's no hint
like hey, buy our new product, just EOL. So let's say I am looking
for an alternative to this. And unfortunately I have to add
- Original Message -
A simply PHP based thing would be OK. Maybe I should look more
specifically for that or can anyone here recommend a PHP based CDR
viewer?
Meanwhile I ended up building a mysql view, for private purposes it
does the job. A real solution would still be nice,
- Original Message -
just wondering if there is any easy to install CDR viewer? Easy
meaning install some package (debian system) and that's it. Had some
problems installing CDR-Stats, FreePBX also seems to be a longer
task for setting up. Isn't there a simple (productive :p)
- Original Message -
- Original Message -
just wondering if there is any easy to install CDR viewer? Easy
meaning install some package (debian system) and that's it. Had
some
problems installing CDR-Stats, FreePBX also seems to be a longer
task for setting up. Isn't
- Original Message -
Hello Ruben,
I belive the problem is not hylafax, is the way dahdi is configure,
here is
a part of the call log:
-- Accepting AUTHENTICATED call from xxx.xx.xx.xx:
requested format = ulaw,
requested prefs = (),
actual format = ulaw,
- Original Message -
Thanks Tim,
I tried your suggestion below the logs:
-- Accepting AUTHENTICATED call from xxx.xx.xx.xx:
requested format = ulaw,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw),
priority = mine
- Original Message -
Yup, there is your problem. Tell hylafax to extend the amount of
time before it times out.
We're a bit off topic for the Asterisk list now, but in your Hylafax
config.ttyIAX0 config file, add this:
ModemWaitTimeCmd: ATS7=120
Restart Hylafax and faxgetty,
- Original Message -
On 07/26/2012 03:32 PM, Danny Nicholas wrote:
Question 1 - I think asterisk only supports a limited set of
statuses
Asterisk does not *receive* presence updates from Polycom phones (or
really, non-Digium phones) at all. Instead, the presence (status)
updates
Another mystery for the list, hopefully someone has ideas on a fix... :)
I've got an Asterisk 1.8.12.0 system connected to a CAS T1 (ESF/B8ZS,
fractional 1-8). Outbound dialing works correctly, but while the call is in
progress, there is no 'ringing' heard by the end user. So, on a SIP phone
Greetings-
I've got a handful of Polycom IP 550 handsets connected to an Asterisk 1.8.12.0
system. Everything is running smoothly with few problems. However, I have an
issue that maybe someone could shed light on...
Many of the phones have 'buddy watch' enabled for the other phones, basically
- Original Message -
Thanks Tim.
One of the problem that I am facing is the complicated generated
configuration for the FreePBX, is it the same thing in the Elastix?
To understand this complicated generated commands, is there a
documentation to explain this for FreePBX or Elastix?
- Original Message -
I'm currently trying to decide on which GUI-enabled version of
Asterisk to use for one particular installation, where we will need
good telecommuter support. We've made it so easy for people to work
remotely that the customer is downsizing their real estate and
- Original Message -
Hello;
Is it possible if I have already asterisk installed on Fedora machine
to install the GUI asterisk now without doing a fresh installation
using the Asterisk Now CD?
Which version of the GUI that should be selected to work with the
asterisk version? For
- Original Message -
It has a Digium Wildcard TE122
If it has an onboard echo canceler, try disabling it and retrying. Just a shot
in the dark, going from my experience with other cards and same symptoms. If
the card is new(ish) I would think Digium could provide support to you for
- Original Message -
OK, is there asterisk-gui that differs that freepbx? Or Freepbx is a
GUI for asterisk?
In other words, if I have asterisk and I need to add for it a GUI, is
there asterisk-gui which is differs than freepbx or it is the same?
There have been a handful of other
- Original Message -
Hi All;
Based on what I have to use Trixbox or FreePBX?
Can someone advise?
Trixbox includes FreePBX as it's GUI. However, keep in mind it is a
bastardized, forked version of FreePBX that has seen nary any new development
or innovation in some time. At this
- Original Message -
Hi Tim,
How about AsteriskNow?
Thanks and BR,
Anam
On 7/7/12, Tim Nelson tnel...@rockbochs.com wrote:
- Original Message -
Hi All;
Based on what I have to use Trixbox or FreePBX?
Can someone advise?
Trixbox includes FreePBX as it's GUI
- Original Message -
I am new. Here is the code that I am playing with on CentOS 6.x
register = 5552530146:funnytiger...@sip3.voipvoip.com
[outgoing]
username=5552530146
type=peer
qualify=yes
secret=funnytiger123
nat=auto
insecure=very
host=69.90.209.57
fromuser=5552530146
- Original Message -
Curiously enough, I can't do that at all on Voip3. Not span 3 of
course, because only span 1 should exist, but I can't execute pri
show spans either.
DING DING DING... we may have a winner. Do you have PRI support on that box,
meaning, did you also compile
- Original Message -
Quoting Tim Nelson tnel...@rockbochs.com:
- Original Message -
Curiously enough, I can't do that at all on Voip3. Not span 3 of
course, because only span 1 should exist, but I can't execute pri
show spans either.
DING DING DING... we may have
- Original Message -
I'm currently running Asterisk 10.5.1, compiled from source, and just
had someone call saying they couldn't get their voice mail. Looking
into the user's voice mail folder, I saw a .lock file.
Removing this file, enabled them to get voice mail.
Is anybody else
- Original Message -
We have the ringer volume issue with some customer environments as
well. We use Grandstream phones in a lot of installs so we just
upload a custom ringtone with the db pushed up on it a bit.
We are testing the Digium phones and have concerns if we will be able
to
- Original Message -
Hello,
1) I am wondering what is the best practice to monitor if there are
or were problems with SIP calls on my Asterisk box. E.g. how about a
software that extracts all calls from the /var/log/asterisk/full (I
have permanently enabled verbose 10 and sip
- Original Message -
- Original Message -
Hello,
1) I am wondering what is the best practice to monitor if there are
or were problems with SIP calls on my Asterisk box. E.g. how about
a
software that extracts all calls from the /var/log/asterisk/full (I
have
Greetings Ron-
Just wanted to give you a heads up about an alternative SCCP channel driver
available for Asterisk. Please see here:
http://freecode.com/projects/chan-sccp-b
I have no experience with it (nor SCCP in general) but just wanted to give you
an option in the event the included SCCP
- Original Message -
Hi Tim,
I'm using Asterisk 10 and on Cisco GW the protocol is set for FAX is
T.38 and when I try to send the fax from a fax machine i.e. HP 3180,
I'm getting some warnings as listed below;
-- Executing [4112345678@default:1] Goto(SIP/192.168.1.69-0005,
- Original Message -
Hi Tim,
...
While the fax machine starts to send the fax after a while it gives
the message, 'Fax failed' with error code: '388'. Is it the end
point fax machine issue or else? Please assist me out to resolve
this issue at earliest.
Please do not email me
- Original Message -
I have installed and configures this card in asterisk 1.6. When
trying to load the module codec_sangoma.so I see the following in
the asterisk log.
[2012-06-04 15:50:31] WARNING[18168] loader.c: Error loading module
'codec_sangoma.so':
- Original Message -
Hi all,
Couple of things I would like ask, does Asterisk provides free
license for FoIP (for 1 channel) or need to purchase it? Couple of
years back, I was able to send and receive the fax using Digium T1
card, in term of FoIP how can I able to receive fax from
- Original Message -
Hi Tim,
Unfortunately i can't reproduce the scenario because it was a long
time ago. But it would be nice to hear from you, what things can be
verified within fax and Asterisk? Any TIP on wireshark monitoring?
Within Asterisk, the debug logs can be helpful for
- Original Message -
Hi Tim,
Thanks for your response. Here is my topology as listing down below;
PSTN Line -- Cisco Voice GW -- IP Cloud -- Asterisk
Will Asterisk able to receive the fax (as in topology above) using
its' fax module? In sip.conf I enabled fax detection and T.38.
- Original Message -
I've used Asterisk 1.4.22 with hylafax and iaxmodem and it wasn't
reliable at all; sometimes the fax reach the destination, sometimes
not, and even worse, asterisk got froozen...(here using analog lines
over Sangoma B600 and Digium TDM400P, same behavior with
- Original Message -
On Tue, May 29, 2012 at 3:10 AM, Danny Dias ing.diasda...@gmail.com
wrote:
Hello,
For those customers with only analog lines, who ask for fax2email
and
email2fax, whats the most reliable solution available and tested
with Asterisk?
Thanks
I've
- Original Message -
On 05/23/2012 08:41 PM, Cody Harris wrote:
Hello All,
I use IAX2 as the incoming connection from my DID provider. For
whatever reason, this works best for me, SIP connections lag very
frequently and only have about a 50% success rate for incoming
calls
- Original Message -
Hi Steve,
you are telling me there is no way to set a particular speed on my
iaxmodem in order to force the sender speed?
I have some problems with a customer who gets malformed faxes even if
no
error occurs. Since I cannot tell the sender to lower its fax
- Original Message -
Hi guys, thanks for answers.
That could seem counter-intuitive but it is not. Not to mention the
fact
that information technology is not science,
Huh? It is indeed very much a science. You have known established facts,
processes, concepts, methods for testing
- Original Message -
On 05/17/2012 07:53 AM, Andrew Furey wrote:
we use ActiveFax for sending (interfaced from an ERP package) and
often get Comm Error 283 and incomplete faxes. If it's just making
a
bad situation worse, how is it that our solution of turning off ECM
mode fixes
- Original Message -
Hi,
I'm facing a strange situation.
Though it's not directly related to Asterisk, I do think it is
interesting to this mailing list.
The setup is a single line which is split between an ADSL
modem/routeur and a fax machine (Asterisk was removed from the
Bart Coninckx wrote:
Hi all,
for smaller (or maybe even bigger) sites I'm looking for a smaller,
appliance-type like PC, preferably solid state and fanless PC.
Since it's only going to run Asterisk for a couple of extensions I
don't think CPU and RAM need to be maxed out.
Does anyone have
- Original Message -
On Thursday 10 May 2012, Bart Coninckx wrote:
I'm looking for a smaller,
appliance-type like PC, preferably solid state and fanless PC.
Since it's only going to run Asterisk for a couple of extensions I
don't
think CPU and RAM need to be maxed out.
Does
- Original Message -
Tim,
looked at these briefly, they all seemed pre-installed, correct? Is
reinstallation with, let's say, CentOS possible?
thx,
BC
The units *can* come preinstalled with our PBX flavor (Debian, Asterisk,
FreePBX), or they can be sent bare and you can
- Original Message -
On 05/10/2012 03:49 AM, Bart Coninckx wrote:
Hi all,
for smaller (or maybe even bigger) sites I'm looking for a smaller,
appliance-type like PC, preferably solid state and fanless PC.
Since it's only going to run Asterisk for a couple of extensions I
don't
- Original Message -
On 05/10/12 18:38, Kevin P. Fleming wrote:
On 05/10/2012 03:49 AM, Bart Coninckx wrote:
Hi all,
for smaller (or maybe even bigger) sites I'm looking for a smaller,
appliance-type like PC, preferably solid state and fanless PC.
Since it's only going to run
Greetings-
I've had reports of a customer PBX acting strangely to some inbound calls.
Specifically, a call comes into an FXO port, hits a Dial() to ring a few
extensions, but by the time someone answers the phone, the call has been
dropped, and the caller is listening to on-hold music. There
- Original Message -
Greetings-
First off, my apologies for the slightly OT nature of this post. It
does involve Asterisk to a degree, but errs a bit on the side of
Audiocodes inquiry. I accept all responsibility for my actions and the
consequences. :)
The scenario is this: I
Greetings-
First off, my apologies for the slightly OT nature of this post. It does
involve Asterisk to a degree, but errs a bit on the side of Audiocodes inquiry.
I accept all responsibility for my actions and the consequences. :)
The scenario is this: I have an Asterisk box connected to a
- Original Message -
Hi,
When someone says T.38 is not reliable on a (normally loaded and
managed) LAN, would you rather agree or disagree ?
In this case, fax calls are coming in through an analog gateway,
passing trough Asterisk and then going out to ISDN through a digital
- Original Message -
Yes, this is exactly what I am looking for - hopefully in English :-)
Date or range selection would make this perfect. I have been looking
for something like this for quite a while but there is none. I would
really appreciate it if you share this with me.
- Original Message -
Hello Folks;
I know this is a non-commercial discussion group, but I am looking for
some open-source software suggestions
We are going to be setting up a prepaid PBX service with the following
features:
• Email to Fax and Fax to Email
• Inward DID
Greetings-
I currently have a customer that *requires* key-system functionality in an
Asterisk PBX. On a SIP phone, the BLF keys need to show the current state of
the analog lines attached to the system (DAHDI FXO). By pressing one of these
keys (for line 1 for example), the dialed number
- Original Message -
On 01/26/2012 09:46 AM, Tim Nelson wrote:
Greetings-
I currently have a customer that *requires* key-system functionality
in an Asterisk PBX. On a SIP phone, the BLF keys need to show the
current state of the analog lines attached to the system (DAHDI
FXO
- Original Message -
I use the latest spandsp source from the freeswitch git.
There you have also a changelog documenting the differences. Steve
Underwood
commit here the latest changes in spandsp source.
http://fisheye.freeswitch.org/changelog/freeswitch.git/libs/spandsp
Does
- Original Message -
I'm trying Asterisk 10.0 (as 8.x is not passing PSTN CallerID) and
Asterisk 10.0 is no better.
I'm still getting:
WARNING[12295]: chan_sip.c:14446 check_auth: username mismatch, have
11, digest has pstn-1270
NOTICE[12295]: chan_sip.c:22769
- Original Message -
On Mon, Dec 26, 2011 at 11:54 PM, virendra bhati virbh...@gmail.com
wrote:
Hi list someone is trying to hack my server . Is there any way by
whcih I can stop hacking of my server except iptables ? I want to stop
on the basis of sip.conf account only. bcoz
- Original Message -
Le 27/12/2011 16:04, Tim Nelson a écrit :
- Original Message -
On Mon, Dec 26, 2011 at 11:54 PM, virendra bhati
virbh...@gmail.com
wrote:
Hi list someone is trying to hack my server . Is there any way by
whcih I can stop hacking of my server
- Original Message -
Hi,
I am new in voip, how many calls can one asterisk box handle with 30 %
of trans-coded calls and system configuration as
8GB RAM
X3430 Xeon Processor, 2.4GHz, 8M Cache, Turbo
This is one of the 'harder' things to calculate. You'll at least want to start
- Original Message -
I spoke with the Asterisk Pre-sales team and they said that SS7
support isn't technically supported, but it is there (e.g. talk to the
OS community about this) so here's my question:
I'm trying to interface an Asterisk Softswitch to a Nortel DMS100.
If I get
Greetings-
On occasion, I'm seeing the following in syslog on some systems using analog
cards with FXS modules:
[ 1664.861183] Power alarm on module 1, resetting!
These are typically cleared by restarting asterisk/dahdi, or power cycling the
system. However, I'm wondering if anyone can
Greetings-
I'm about to dive into the process of virtualizing some of my Asterisk
(primarily 1.4.x) infrastructure. In the past, when looking at virt solutions,
the primary issue preventing me from moving was the lack of proper timing. We
do not need it for MeetMe but rather for IAX2 trunking.
Greetings-
I'm working on a unique Asterisk installation where I've been given a
requirement of keeping a voice call active, even during a data connectivity
loss. So, let's assume I have remote users connecting to an Asterisk server via
sometimes unreliable connectivity such as satellite,
Greetings-
From time to time, I find myself working with (or customers working with)
dynamic T1s. They are typically standard T1s that terminate to an Adtran
device which utilizes the channels for data (64kbps X 24) until a call is
pushed inbound/outbound on the circuit. One data channel is
- Original Message -
On 11-09-01 07:04 AM, Tim King wrote:
I have found numerous claims that 1.8 can do T.38 gateway with a
patch,
however I am yet to find the patch or any instructions on
implementing it.
Anyone have a link?
Asterisk-10.0.0-beta1 is another option.
I've
- Original Message -
Well, we've taken the time to check out the wiring. It's only 3 years
old and
looks like the people who did it knew what they were doing. Nice work.
Rebooting the cable modem, router, and switch didn't fix the problem.
Also, we had an instance today where ALL
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