Caller ID information in a Remote-Party-ID (or
P-Asserted-Identity, depending on the version you are using) header,
allowing the From header to be used solely for authentication.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@di
:
SetGlobalVar(VOLUME(TX)=10)
SetGlobalVar(VOLUME(RX)=10)
Dialplan functions cannot be set globally.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
soeren
Hi Soeren,
Yes, "module show" shows all three of them.
Please check to see if there is an issue open for this problem on
https://issues.asterisk.org/jira. If there is not, please open one; an
incorrectly formatted configuration file should not result in a segfault.
--
Ke
)
You are stopping the Asterisk SIP channel driver from doing its job; it
expects the channel to be dead much sooner than 25 seconds after
receiving (or sending) a BYE. Why do you need to keep the channel alive
for so long after it has been hungup?
--
Kevin P. Fleming
Digium, Inc. | Director o
d attended transfer' though, because that doesn't really make
any sense. Whether chan_sip will use REFER with a Replaces header or not
to effect the transfer I can't say for sure, but it will cause a blind
transfer of the channel to the destination specified.
--
Kevin P. Fleming
D
ludes the patch in question, then RPMs and DEBs don't have it either.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
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Check us out at www.
u can use the SRPM for Asterisk to rebuild the RPM after importing the
iLBC source into the build tree; at least I think that would work.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis
On 08/01/2011 03:35 PM, Paul Belanger wrote:
On 11-08-01 04:24 PM, Daniel - Asterisk wrote:
You are closing the socket before reading the result of 'Logoff' and
Asterisk is complaining.
Well, he's sending DBPut before reading the result of Login as well.
--
Kevin P. Flem
ssage subject :-)
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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Check us out at www.digium.com & www.ast
codec is nearing completion, and it is very likely that it
will be incorporated into the WebRTC stack soon after that. Given that,
there's not much reason to spend time working on ISAC.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: k
han->writeformat
3. chan ->rawreadformat
4. chan ->rawwriteformat
5. chan->nativeformats
Code questions should be posted to the asterisk-dev mailing list.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype
to narrow down the bug.
If it was a regression from 1.6.2.18 to 1.6.2.19, then it will be fixed.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check
l drivers and
using control frames that pass through bridges. It would be a large
amount of effort to implement it again in 1.4/1.6. It extends well
beyond simple dialing, as it can receive updates across external
protocols and pass them along, it handles call redirection, and various
other fe
has been completed to show who the person is talking to (not the person
who performed the transfer).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
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d ensure you are running 3.x software on
the phones) to find out if those features can be disabled.
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Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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Chec
much
larger, but we had to remove some because they sound terrible when
compressed with G.729.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 -
9 file, or should I just give up and
change everyone to ulaw ?
G.729 is a *speech* codec, and as such it does not handle non-speech
(music, tones, etc.) very well at all.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.c
er respond with that code in
that situation.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digi
).
Alternatively, you could schedule a manual power outage and determine
*why* outbound calls fail after the power returns, so that you don't
need to reboot at all to get them working. Address the cause, not the
symptom.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
J
TIFY,REFER,OPTIONS
Content-Length: 0
Nay body know what's wrong here ?
What makes you think something is wrong? Nothing is wrong here, this is
perfectly normal.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | S
mbination has a SIP registrar listening on it, they'll
attack it.
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Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
f the asterisk-sounds-moh RPMs installed?
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.ast
On 07/21/2011 04:43 PM, Israel Gottlieb wrote:
On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming mailto:kpflem...@digium.com>> wrote:
On 07/21/2011 04:34 PM, Joaquin Sosa wrote:
On Mon, Jul 18, 2011 at 07:58, Steve Daviesmailto:davies...@gmail.com>> wrote:
ementations from more than one vendor is (unfortunately) likely to
have problems, whether any version of Asterisk is involved or not.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive
Sonus device sent "fmtp:101 0-15" in its SDP, then Asterisk
should not send 'event 16' events to it. If it does, that's a bug,
although standard programming practices would mean that it wouldn't be
harmful, it would just be ignored by the Sonus device.
--
Kevin P. F
t codes 0 through 16, but the other endpoint is not
obligated to send them if it doesn't want to.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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ior to 1.8.3.0 and I
do not see any issues in /var/log/asterisk/messages ?
No, this is not expected behavior.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL
ingle Asterisk dialplan,
just via different ports).
The lightest weight solution for this problem is a stateless SIP proxy.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsvill
h for
pbx_builtin_setvar_helper() function calls where the variable name
starts with "SS7_". If you have more specific questions about Asterisk's
support for SS7, join the asterisk-ss7 mailing list and ask there.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Techno
incompatible signalling?
They are completely incompatible above the physical and HDLC layers.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out
On 07/19/2011 01:16 PM, Alex Balashov wrote:
On 07/19/2011 02:15 PM, Kevin P. Fleming wrote:
Actually, you can do this with one installation of Asterisk, and a
separate set of config files and data directories. When the Asterisk
executable is started, the '-C' option can be used to p
g one of the init scripts, then yes, that would need to
be duplicated and modified.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806
e 2nd instance to listen on another port?
It would be much easier to install a SIP proxy to listen on the second
port and forward requests over to Asterisk on the standard port.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digiu
call that sip user, both sip clients will ring.
No, it's not. Asterisk does not support multiple registrations to the
same SIP AoR.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis
CDR modules for PostgreSQL and FreeTDS (Microsoft SQL Server),
and also generic ODBC support which can be used to connect to MySQL if
you wish.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan
On 07/18/2011 05:05 PM, Elliot Murdock wrote:
I am wondering if the Libss7 add-on for Asterisk also translates ss7
variables into the dialplans for routing, accounting, etc?
What are 'ss7 variables'?
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jab
duced in 1.6.2.19, then it should still be fixed.
At least I believe that's the rules.
That should be the case, yes.
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Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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me to get RPMs properly built and tested.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.ast
ution time is all that's important, right?
OT: Take a look at 'systemd'; this is exactly what's happening there,
and Fedora is likely to incorporate it into Fedora 16, and it will make
its way into other distros after that.
--
Kevin P. Fleming
Digium, Inc. | Director of Sof
sing the network any more. It would be best to
plan for it being non-functional after the two year support period is over.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
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ne can't decide when a call is 'not answered'. However,
writing such a dialplan would indeed be non-trivial :-)
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Dri
er the call ,A and B Call should connect back.
IMHO, blind tranfer definition is to NOT connect A and B back
That is correct, and is why it's called a 'blind' transfer; the
transferring party does not know or care what happens to the call after
effecting the transfer.
--
Kevin P
e that you need to force G.711 to be used?
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www
lternatively, if your provider claims to support T.38, then ensure you
are running an updated version of Asterisk, and if you still can't make
FAX work over T.38 with them, post debug logs here and we can try to
help you figure why it's not working.
--
Kevin P. Fleming
Digium, Inc. |
outgoing FAXes (and redirect the incoming channel
to a different destination in the dialplan). Now that chan_sip has
'faxdetect' as well, many usages of 'outgoing' in chan_dahdi are no
longer necessary.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabb
le (but not
included with Asterisk) to do that. I believe NVFaxDetect can do it, and
most of the add-on answering machine detection applications can do it as
well.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skyp
ywhere). Your message had
three lines of content and 30+ lines of non-content.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digi
just drop everything and fix it
when Google changes the protocol.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & ww
nd documented in the manual for the Hx8 cards:
change the 'te' in the 'span' line to 'nt'.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
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d and the
appropriate permissions granted to the user.
What is probably happening here is that Safari does not handle the
'optional' client certificate request from the server properly.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.co
"SSLCertificateChainFile /full/path/to/your.ca-bundle"
Can Safari open a connection to https://issues.asterisk.org? (no /jira
suffix)
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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445
.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk
stuff Switchvox isn't
really designed for.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & ww
in Asterisk 1.8 is very different from the one in
trunk (what will become Asterisk 1.10).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us
se
branch. 1.4.40.2 was released so that 1.4.40/1.4.40.1 users could get a
security fix regression resolved without having to move to 1.4.41 if
they are not ready to do so.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.c
st) for anyone to be able to determine what might
be happening.
The quick answer, though, is that Asterisk will use whatever payload
number for RFC2833 DTMF that the other end requests. The message you are
seeing has nothing to do with DTMF.
--
Kevin P. Fleming
Digium, Inc. | Director of Software
On 04/20/2011 04:55 AM, Niccolò Belli wrote:
Il 19/04/2011 23:41, Kevin P. Fleming ha scritto:
If you are the receiver of the call (and thus they are the sender of the
call), it is *your* system's responsibility to initiate the switch to
T.38, not theirs.
Are you sure? So what's fax
number for incoming FAX calls and not rely on
'faxdetect' at all; this would allow you to use G.729 for your voice calls.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
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ri is used only in userspace, and has nothing to
do with anything in the kernel.
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Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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Check us out
for parameters to be passed to a module.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
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Check us out at www.digium.com & www.asterisk
#x27; for all of these items, which means you *must*
have them installed. Have you made any changes to the Asterisk source code?
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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heck for them, and see what is failing.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www
se go read RFC 2833 or RFC 4733; they explain how digits are sent
over RTP.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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Check us out at www.digiu
ndicate what format they contain. Have you opened the
files with Wireshark or any other tool that can interpret PCAP files?
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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ader.c:777 load_resource: Module
'app_voicemail_imapstorage.so' could not be loaded.
Is there some way to have this working?
Yes... but this indicates that the module that was built appears to be
broken. I'll let the package maintainer know.
--
Kevin P. Fleming
Digium, In
I wouldn't be concerned about that specifically.
Given the fact that the phone is not incrementing it's OSeqNo in the
REGREQ packets you showed in the capture, I would agree that it appears
that the replies from Asterisk are not being received by the phone.
--
Kevin P. Fleming
address.
Worst case, use tcpdump to make a packet capture of the traffic to/from
the phone and then use Wireshark to look at what is going on.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan
or FAX call, in which case
it could cause the call to fail).
For your sanity, I would strongly suggest that you don't connect spans
from multiple telcos/networks/etc. on a single card, but keep each span
provider on their own card.
--
Kevin P. Fleming
Digium, Inc. | Director of Software
and make a
Local/234@somecontext which dials SIP/234-foo&SIP/234-bar.
Why do you need a Local channel to do this? If extension 234 exists in
some context, the Dial() statement in that extension can dial
SIP/234-foo and SIP/234-bar itself.
--
Kevin P. Fleming
Digium, Inc. | Director of Softwar
On 03/15/2011 04:18 AM, Nikhil wrote:
how to send SIP HOLD Invite from asterisk to other sip client/server.?
Asterisk's chan_sip does not yet have the ability to *send* 'hold'
re-INVITEs.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@d
locks to occur, and the cause has not yet been found.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.ast
/h', and he had no peer named 'h' or is that an
IP address or DNS name. It should have failed a little more cleanly than
it did, but I'm sure that at least part of the problem is attempting to
dial a SIP endpoint that doesn't exist (and dialing out from the 'h'
r way, 200 phones answering a call at the same instant is a *lot*
for Asterisk to handle. This is why multicast paging is preferred, but
as others have pointed out, it doesn't appear that Polycom phones
support that type of paging.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Tech
1.8 has a built-in Page() application you can use
from the dialplan to achieve what it appears you were trying to achieve
with your AGI script.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan
ou. For example, 'sip show
peer' output confirming that *both* SIP endpoints have T.38 enabled.
Then, a complete 'sip set debug on', 'core set verbose 10' and 'core set
debug 10' console capture of a failing call, so that we can see what
happened with the T
nswer to all of those questions is probably 'yes', but that's why I
said someone with SPARC experience would have to chime in.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
44
On 03/07/2011 04:31 PM, RR wrote:
On Mon, Mar 7, 2011 at 5:25 PM, Kevin P. Fleming mailto:kpflem...@digium.com>> wrote:
Please do not reply directly to posters on the mailing list unless they
request it.
On 03/07/2011 03:35 PM, RR wrote:
Hello all,
mmm
t any echo problem.
Where do I start to figure this out? How do I narrow it down? Can I
figure out if it is an iaxagent problem? Could using jitterbuffer cause
this?
This is probably acoustic echo from your phone. The jitterbuffer has
nothing to do with this.
--
Kevin P. Fleming
Digium
s like something is selected
that doesn't have other related stuff unselected? no clue where to start
looking
Have you specified any '-march' or '-mcpu' options to the compiler? This
sort of thing can occur if you are building for a plain-jane i386
processor or something si
[2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: Copyright (C)
1998-2008 The OpenSSL Project/
How can I fix this WARNING error?
You can follow the instructions with the product and ensure that
res_fax.so is loaded before res_fax_digium.so.
--
Kevin P. Fleming
Digium, Inc. | Director of Software
generated these logs. I'm
not sure if this is a chan_sip.c problem or if this is a dial plan problem.
If your version string is 'SVN-trunk-r309404', you are not using 1.8,
you are using 'trunk'. If you want to follow the 1.8 Subversion branch,
you need to checkout that br
y aware of your server and
customers) can spoof the IP addresses of your server(s) in order to get
the remote endpoints to at least accept an INVITE (they can't place a
successful call through them using spoofing though).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
44
ere are quite
a few and their behavior in this situation could be different.
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Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfl
Cisco.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com & www.asterisk
phone
(which it can optionally generate for DND being enabled and disabled).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com & www.ast
rd drive
from someone who isn't supposed to have access to it, not the system's
normal user.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at w
point, although there are some
pretty creative people out there, so who knows :-)
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com & www.ast
takes rather longer to figure out a way
around the obscuring mechanism(s), but if enough people are interested
in doing so, they will. With open source software, pretty much anyone
can get around such mechanisms in a short period of time.
--
Kevin P. Fleming
Digium, Inc. | Director of Softw
ing also distributing the modified source code,
and thus the same problem arises.
"Security through obscurity" does not work with open source software.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | j
an different
things in a single sentence :-) Clarity and completeness make it much
easier for people to understand what you are trying to express.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfl
sword can be un-done by a motivated person. The only question is
their level of motivation :-)
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digiu
s obscured passwords using 'md5secret', but all
other protocols that Asterisk supports need the password in plaintext to
be able to perform the authentication process required by that protocol.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive
extracting these passwords.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com & www.asterisk
(which I think is what you're expecting).
This is correct. 'reload' is not 'restart', it only tells all the
currently-loaded modules to 'reload' themselves (which generally means
they will reparse their configuration files to look for changes).
--
Ke
a PRI.
Asterisk does not currently support T.38<->TDM gateway mode for FAX,
although there is a patch on the issue tracker to add support for it,
and it's in the works for Asterisk 1.10.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW -
or the card; if you didn't receive a printed
copy when you purchased it, you can read it online on www.digium.com.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us o
ue for someone to look at.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digi
s the
same, but the source of the problem is quite different.
This can of course cause complications if Dial() is used to dial
multiple endpoints... because then there could be multiple audio streams
received from them as the call proceeds towards one of them answering.
--
Kevin P. Flem
Asterisk trunk after 1.6.2
was branched (so only 1.8.0 and trunk are missing the code). Leif Madsen
entered an issue on Mantis as a blocker for any more 1.8.x releases
until this is resolved, as it is clearly a regression in the 1.8.x series.
--
Kevin P. Fleming
Digium, Inc. | Direc
n Asterisk since March of 2010.
If I had to guess, though, I would think that that receiver of this call
has sent your system a T.38 re-INVITE before you started SendFAX
running. If that is the case, this problem has already been solved and
upgrading will take care of it.
--
Kevin P. Flem
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