, because
the fax for asterisk admin manual, there are no
information about the T.38 error correction, and if i better use
Redundancy or FEC.
Please contact Digium Support with questions about Fax For Asterisk's
operations and features. Thanks.
--
Kevin P. Fleming
Digium, Inc. | Director of Software
usage. This is what GR-303 was designed (and is still used) for.
--
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.org
to random
scheduling-related problems.
--
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.org
it in production with these errors.
The message is labeled WARNING, which means it is not an error. This can
be ignored, unless you are actually experiencing a problem.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming
' extension is running, the call legs have already been torn
down. There is no way to delay this happening, and you can't do anything
in the 'h' extension that needs to read audio from the channel (since no
audio will appear, the first time it tries to read audio it will abort).
--
Kevin P. Fleming
are those that don't have anything to do with the external channel that
was involved before the hangup. No audio, no DTMF, etc.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
... and if at any time waiting for or reading media
from the channel fails, it exits, because there's no point in continuing
to wait since the call is gone.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming
?
The version we are running is DAHDI 2.2.1, Asterisk 1.6.0.22
Those messages have always been present since Zaptel. It means that
the port detected a Fax signal and is disabling echo cancellation.
And it's not an error, so there's no need to do anything about it.
--
Kevin P. Fleming
be one of many different versions, and could potentially have
significant patches applied... which makes it more difficult for the
provider to be comfortable that it will 'just work'.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806
it, which would
absorb these ring splashes.
--
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.org
, and each one
will match one of the branches of this tree, and it's values will be
extracted and stored for later use.
In other words... this is not the cause of your problem.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
it handle the SIP/TLS - SIP/UDP
conversion.
--
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.org
: No database host found, using localhost via socket.
WARNING[1819]: res_config_pgsql.c:1383 parse_config: PostgreSQL
RealTime: No database port found, using 5432 as default.
But there is no connection being made to the database.
What version of Asterisk are you using?
--
Kevin P. Fleming
On 05/22/2010 09:22 AM, Deepesh D wrote:
I am using Asterisk 1.6.2.7
On Sat, May 22, 2010 at 7:20 PM, Kevin P. Fleming kpflem...@digium.com
wrote:
On 05/22/2010 02:07 AM, Deepesh D wrote:
I tried removing the dbhost and dbport entries and restarting asterisk.
During startup
*both* a socket to be used and a hostname/port number.
The way the code is written, if both are supplied, the host/port
combination is used and the socket path is ignored. If you don't want
the host/port to be used, don't specify them.
--
Kevin P. Fleming
Digium, Inc. | Director of Software
will *always* accept properly formed re-INVITEs that don't
require capabilities that are not available, and it will also generate
them for non-directmedia purposes (like switching to and from T.38) when
necessary, regardless of whether 'canreinvite' is set to yes or no.
--
Kevin P. Fleming
Digium, Inc
the
channel is still on hold. Are you using 'mohinterpret=passthrough',
where Asterisk would send the hold indication to the bridged channel
instead of reacting to it locally?
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
, it's a known problem, and the fix should be out within a
day or two. It was reported to us about a week ago, so if you had
contacted the support department, it's likely they would have been able
to shortcut your hair-pulling experience :-)
--
Kevin P. Fleming
Digium, Inc. | Director of Software
it
wouldn't make a very good device to provide FAX termination and
origination without some work on the web interface.
--
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
)
that do this; look for OrecX.
--
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.org
are supported?
No video codecs are supported; Skype clients only support VP7 and H.264
(most of them VP7), so it's not clear what is going to be possible once
SFA does have video support.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806
for a
channel driver like chan_skype, so it must be distributed as source code
and compiled against the configured and installed copy of Asterisk.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem
Which line is 'line 23' of the T.38 re-INVITE?
--
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.org
advantage of it... hopefully also this week.
--
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.org
.
--
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.org
forward media between the two sessions.
--
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.org
that itself.
--
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.org
is
made for an updated Skype For Asterisk release to go along with it. If
it does occur using the latest release, then you should contact Digium
Support to report the issue so it can be expedited to the Skype For
Asterisk maintainers.
--
Kevin P. Fleming
Digium, Inc. | Director of Software
environment.
RT kernels don't have traditional mutexes, which are used in various
places in DAHDI for Linux. To my knowledge nobody has done the work to
update the drivers to be able to use the RT kernel replacement
synchronization mechanisms when compiled against an RT kernel.
--
Kevin P. Fleming
on the used
market.
--
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.org
expensive PBXes.
--
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.org
versions.
--
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.org
where app_voicemail spit out prompt file names as the 'connected
party ID', for example.
--
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
Olivier wrote:
Is it me or is svn.asterisk.org http://svn.asterisk.org down ?
It is, along with issues.asterisk.org, reviewboard.asterisk.org and some
other sites. They should be back up in the next hour.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive
calls, the calls will be
processed through your dialplan, and you can forward them on to the E1.
There's no need to 'allocate' bandwidth to this device, it will ask for
what it needs when it needs it.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW
, that's unfortunate, and I agree that we
need to get that information posted somewhere so that when someone
searches for it they'd be likely to find it.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber
.
--
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.org
--
_
-- Bandwidth
in that building have
been using the 1 phone and thier cell phones for a few days now - but I
really need to find a fix for this.
Since you bought the product from Pika, you should call Pika.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville
.
--
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.org
received RTCP from server.
I use Asterisk 1.6.2.6 or 1.4.29 .
Also SIP/RTP.
There is no configuration option for doing this; RTCP is a mandatory
part of an RTP implementation that intends to be compliant with the
RFCs. If you want to disable it, you'd have to modify the source code.
--
Kevin P
).
If you can suggest a method to provide this information to people in
some automatic way when they are made aware of the conflict by RPM, feel
free to do so and we'll try to get it incorporated into the RPMs themselves.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan
.
This is called 'Connected Party ID', and it isn't supported in any
released version of Asterisk... but it is supported in SVN trunk and
will be part of Asterisk 1.8.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype
, but it has been that way forever so we can't change
it. The [general] section *should* have only been for settings that
apply to the SIP channel driver as a whole, and *not* for providing
defaults to entities configured for the driver. Unfortunately, it has
both purposes.
--
Kevin P. Fleming
Digium, Inc
, and
not defaults.
--
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.org
in the CHANGES file.
--
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.org
on your expected
traffic, and also used overlapping ranges, it would be easy for calls to
fail because there are no port numbers available. Using non-overlapping
ranges will make this much less likely.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW
send; there are also changes being made in the
SkypeIn and SkypeOut networks to properly support DTMF. Stay tuned :-)
--
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
the expiration timer to 'zero'.
--
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.org
and
udptl.conf)
It absolutely would be a problem to have identical, or even overlapping,
port ranges specified in rtp.conf and udptl.conf. Those port numbers are
UDP port numbers, and they must be unique across the system for things
to work properly.
--
Kevin P. Fleming
Digium, Inc. | Director
'opinions' (the 'O' in 'MOS'). However, it
seems that many people use PESQ scores as a MOS-equivalent for test and
planning purposes now. However, that requires running predefined samples
through the system under test, not just calculations based on network
effects of real calls.
--
Kevin P. Fleming
cause chan_dahdi to go off
hook, skip sending any digits, and go into 'answered' mode.
--
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
of thing is easy to do using ExternalIVR instead of AGI.
--
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.org
Tzafrir Cohen wrote:
http://downloads.asterisk.org/pub/security/AST-2009-006.html
http://downloads.asterisk.org/pub/security/IAX2-security.html
And more importantly, the UPGRADE files included in the source code that
the OP downloaded pointed to all of this stuff.
--
Kevin P. Fleming
Digium
the dialplan contains any steps to be done with B's channel before it is
destroyed.
--
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.org
. This way people
who find that thread in the list archives can see all the messages in
the thread. Thanks.
To answer your question, though, no, there is no method available in
Asterisk today to modify this behavior. Are you just curious, or do
think it is actually causing a problem?
--
Kevin P. Fleming
at all, this would be possible, but for a B2BUA like Asterisk, it's not
likely to be possible.
--
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
getting dropped if an RTP timeout is in use.
--
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.org
available. MeetMe not only requires a timer, the mixing itself is done
in DAHDI/Zaptel, whereas ConfBridge does the mixing in the application.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem
header content from within the dialplan?
The SIP_HEADER, SIPAddHeader and SIPRemoveHeader dialplan functions
should do exactly what you want.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem
this away)*
Calls don't have event headers; Event packages are used for
subscriptions.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com
be done if it was deemed useful and worthwhile.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com www.asterisk.org
if you loose one packet
sent right now you probably loose all of them.
Presumably then for FEC/redundancy purposes you treat this as if the
application had delivered 'n' copies of the IFP as well.
Spacing them out in time could be complex, to say the least :-)
--
Kevin P. Fleming
Digium, Inc
applications
like Asterisk.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com www.asterisk.org
sees 'globals'. The simple fix for this is to put an
empty [globals] at the very top of extensions.conf, then in any included
files (and later in extensions.conf), use the (+) syntax.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806
instance of Dial and doing so would just result in an
infinite loop.
You need to figure out why the device at SIP/100 told Asterisk to
forward the call when you were expecting it to just accept it.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW
.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com www.asterisk.org
Karl Fife wrote:
Down for me too.
It's fixed now, sorry for the disruption.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com www.asterisk.org
be removed, and the
test tried again.
In order to test that quickly, you could edit apps/app_fax.c and find
the lines that set transcoding_mmr and transcoding_jbig to '1', and
change them to '0' (zero) or remove them.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan
to the
various branches it belongs in. I'd still like to hear from Steve
Underwood if I misinterpreted the MMR/JBIG transcoding function calls in
spandsp that led me to enabling these features in the first place...
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW
of T38FaxTranscodingMMR and T38FaxTranscodingJBIG
from app_fax because of the presence of the
t38_set_mmr/jbig_transcoding() calls in the spandsp API. I will admit to
not reading the documentation to see if they actually did anything
useful, though... should I remove them?
--
Kevin P. Fleming
Digium, Inc
.
Don't know the nuances, but the thresholds are definitely configurable.
By default OpenVPN runs over UDP, and does not provide any guarantee of
delivery at all. It acts just like a datagram delivery protocol, like
Ethernet or any other layer 1 protocol.
--
Kevin P. Fleming
Digium, Inc
of any plans to produce them.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com www.asterisk.org
as 'capabilities' (a=cdsc and a=cpar) which Asterisk does not
support. The second example does not provide backwards compatibility for
SIP endpoints that do not support capability-based negotiation, whereas
the first one does.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan
Steve Underwood wrote:
Hi Kevin,
On 02/02/2010 09:12 PM, Kevin P. Fleming wrote:
VinÃcius Fontes wrote:
[Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing
session-level SDP v=0... UNSUPPORTED.
[Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp
in
clarifying the situation.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com www.asterisk.org
; that
will make it much more obvious what is happening.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com www.asterisk.org
using app_fax/spandsp, I believe we
currently have app_fax configured to offer TranscodingMMR and
TranscodingJBIG because spandsp supports those modes. The Fax for
Asterisk product does not, so re-INVITEs generated by that
implementation would not include those options.
--
Kevin P. Fleming
Digium
For Asterisk, as it does not generate any audio frames while
negotiating T.38 as the receiver of a FAX.
I would suggest opening an issue in the issue tracker at
issues.asterisk.org and uploading your console trace there; there is
clearly a bug here that needs to be found and fixed.
--
Kevin P
was not able to install it with version 1.4
It has never been offered for inclusion into Asterisk, so they cannot
do it. The Asterisk project does not 'pull in' code, it must be
specifically offered for inclusion by the author(s)/copyright holder(s)
of the code.
--
Kevin P. Fleming
Digium, Inc
we're just part of a cluster handling a domain or if we're THE
domain handler.
That would be a good idea, yes.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out
machines; it is for detection of calling FAX machines.
The open source NVFaxDetect application may be able to do what you want.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Szasz Szabolcs wrote:
How can I disable comfort noise on Asterisk?
Asterisk does not have a comfort noise generator, so there is nothing to
disable. You'll have to be more specific about what you are trying to
accomplish.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
as a different entity, which is a different scenario.
It is very likely that there is no standard-defined 4xx code for 'cannot
process this call right now', only the 5xx and 6xx variants.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806
. And even
if they do, they have to know the message is there to seek on the recording.
In the US at least, calls to PSAPs are recorded from the instant the
last digit is dialed, before the call is even routed and ringing (on
wireline networks where this is possible, anyway).
--
Kevin P. Fleming
to make
it work 'better' :-)
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com www.asterisk.org
easier, use 'dahdi show channel X' and see if faxdetect is
indeed disabled.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com www.asterisk.org
JR Richardson wrote:
Hi All,
I'm using Asterisk 1.4 branch and checking the status of some SIP
Peers with the functions ${SIPPEER(101:status)} and the result is OK
(48 ms). Seems to work fine.
That is a bug; the function should be returning OK without the
calculated lag value.
--
Kevin P
in the sample
features.conf it describes exactly how to enable the features you have
defined in that category.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out
:
load = app_fax.so
Rebooted. No module loaded:
# lsmod | grep fax
#
app_fax is not a kernel module, it's an Asterisk module. 'lsmod' is
never going to show it.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806
mean like 'module show'? Or 'module show app_fax.so'? Those commands
already exist.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com www.asterisk.org
responses for that INVITE.
You need to not answer the call until you really want to answer it and
keep it.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out
on
'intelligent signaling' channels like SIP and ISDN; there are nearly
always other, proper, ways to get the desired effect.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check
jonas kellens wrote:
So if I use early media (not putting answer() at the beginning of my
dialplan), how can I send a 503 or 603 from the dialplan ??
By using the proper method of canceling the call... Busy, Congestion, or
an explicit cause code passed to Hangup.
--
Kevin P. Fleming
Digium
or forward. Note
for the list admin: Please preceed your message-footer with a sigdashes
line!
Good idea, done!
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out
codecs/codec_lpc10.c' before running the 'make' and 'make install'
steps.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com www.asterisk.org
that common? I'd be surprised if an endpoint would want
to consume a G.729 encoder (for example) without a corresponding decoder
on the receive path... doing that would make managing DSP resources in
the endpoint much more complicated.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
what you want.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com www.asterisk.org
___
-- Bandwidth
that it is not also willing to receive; in fact, I
can't say I've ever seen this situation arise in any testing I've done
or in any issues reported in our issue tracker.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype
for
this to work, or provide a wrapper for pppd that ZapRAS can execute with
the suid bit set so that pppd runs with root privileges.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check
dialplan will halt
execution until PPPD returns, so there's no way you are going to be able
to execute an AGI or System() or anything to take actions over the PPP link.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype
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