reting the RFC. IAX2 cannot use a separate
signaling and media stream to setup a call, but it *can* optimize a
media stream for a bridged call so that the media does not have to make
as many hops as the signaling does. The media still moves on the same
ports as the signaling packets, using th
end they don't
affect how Asterisk operates, only the speed at which it does so.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfl
ve any effect is if you compiled binaries specifically for one family
of processors and used them on the other. As far as how the software
operates, by definition the processor type/family does not matter at all.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Dr
same system.
--
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 08/05/2010 06:25 PM, Roderick A. Anderson wrote:
> Kevin P. Fleming wrote:
>> On 08/05/2010 03:52 PM, Roderick A. Anderson wrote:
>>> I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6
>>> installed from the asterisk.org and digium.com repositori
in your
modules.conf file. What packages have you installed from the
asterisk.org and digium.com yum repositories?
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: k
On 08/02/2010 02:34 AM, Siju George wrote:
> Hi,
>
> Is there any Free software that can connect to an Asterisk Server and
> Do video Conferencing? or atleast one to one video chat?
One to one video chat is already supported by Asterisk, using SIP or
H.323 video phones.
--
Kevi
On 07/28/2010 08:20 PM, Landy Landy wrote:
> Jeremy,
>
> Thanks a lot that helped and solved the problem. I had it as:
> voice=Marta-8kHz before and that didn't work and now changed it to
> voice=Marta.
That's because you only have the Marta-16kHz voice installed.
--
ed server environments,
or in virtualized environments, that a hardware device providing timing
might be able to maintain proper timing better than DAHDI core timing
can, but there's no way to know that without testing the specific
environment.
--
Kevin P. Fleming
Digium, Inc. | Direc
lready, you could switch to using the
Adhearsion framework, which makes interaction with Asterisk trivially
easy, and handles all the AGI/AMI stuff 'under the covers' for you.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 -
ng, which is unfortunate, but it's
also a way to reduce the burden on our development team during the beta
testing period.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@di
#x27;/n' option to keep it in the middle of the path, which would keep the
entire call inside Asterisk and simplify the configuration a bit.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpf
ensions and place the phones into a
> MeetMe/Conference bridge.
This is exactly what the Page() application does, as has already been
pointed out in this thread. No need to reinvent this wheel :-)
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive N
On 07/23/2010 04:40 PM, bruce bruce wrote:
> You can also use Ethernet Over Power Lines solution or wireless :-)
His issue wasn't getting the network connection delivered, it was the
power :-)
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis
earch for "PoE extractor". Here's
an example:
http://www.shireeninc.com/poe-extractor.html
--
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.
>
>
> I have no control over my firewall – send to da...@debsinc.com
> <mailto:da...@debsinc.com>. Thanks
>
The file in question is probably part of Flash Operator Panel, in which
case it is readily available in many other places on the Internet already.
--
Kevin P
d
conferences, although there are some being worked on (XCON in the IETF,
for example), so right now with SIP phones you are limited to the number
of channels the phone can mix itself if the phone is managing the
conference.
--
Kevin P. Fleming
Digium, Inc. |
e suggestions for Windows
> -- OSS if possible, but payware is acceptable.
In addition to the suggestions of Zoiper, there is also Blink, although
their primary version is on OSX and the Linux/Windows versions are just
now arriving in early releases.
--
Kevin P. Fleming
Digium, Inc. | Di
hat won't work either, because a WAV file has a header, and a raw
alaw file does not... so Asterisk will try to play the contents of that
header as alaw data, presumably producing terrible noise.
The best you can do is to use sox to convert them from
alaw-in-WAV-container to raw-alaw
original poster: all of Digium's hardware echocan products are
compliant with G.165 and G.168 for tone detection and either completely
disabling the LEC or just disabling the NLP portion, depending on which
tone is detected. CED detectors will typically detect ANSam as if it was
CED, which
ber, instead of the
actual channel, and then do your logic in the context/extension you
specified before performing the actual dial operation.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@
Kit is basically a headless Skype client.
SkypeKit is currently single-user and single-call, just like the regular
Skype client.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kp
n hold, but there cannot be
more than one active call). If this suits your needs, you can certainly
try it. There are other Skype gateway solutions that use a similar
method, but they are not free.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW -
and 'C', then *don't* setup SFA to allow calls from anyone, and
>> don't set it up to automatically add users to the buddy list when they
>> request it. Instead, manually add users B and C to A's buddy list
> (using
>> a regular Skype client), and thos
'A', and your remote users are
'B' and 'C', then *don't* setup SFA to allow calls from anyone, and
don't set it up to automatically add users to the buddy list when they
request it. Instead, manually add users B and C to A's buddy list (using
a
commercial Fax For Asterix is free of that problem?
You are already using 'commercial' Fax for Asterisk (not Asterix).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.c
already been fixed in recent releases of FFA; there was a bug
previously where the module would cause Asterisk to crash if a document
to be sent could not be queued (for one of many reasons).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsvil
f SIP requests and responses. Asterisk is a B2BUA UA, so
the two SIP dialogs involved in a 'call' are completely separate.
Asterisk does not have any support for 100rel or PRACK.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806
ies once again and please do not reply.
Threads cannot be deleted from the list; once messages are posted, they
appear in the archives (of which there are many) and are delivered to
thousands of subscribers. Sorry.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Dri
seriousness though, is there not a way to detect
> this behavior and handle the answer() correctly?
The Dial() application can already play an announcement to the called
party and wait for them to confirm the call before accepting that the
outbound channel is 'answered'. This allows
On 07/07/2010 03:33 PM, Tilghman Lesher wrote:
> On Wednesday 07 July 2010 14:58:05 Kevin P. Fleming wrote:
>> On 07/07/2010 10:52 AM, Tilghman Lesher wrote:
>>> On Wednesday 07 July 2010 05:24:10 A J Stiles wrote:
>>>> On Tuesday 06 Jul 2010, ABBAS SHAKEE
ster deciding to modify Asterisk to decrypt files as it reads
them... and even then, the license violation would only occur if they
failed to provide their customers the modified Asterisk code; keeping
the decryption keys private would not violate the GPLv2 at all.
How does obtaining a commercial
ts the
desired iax.conf content for the server it is running on. That's much
easier and more effective than trying to put conditional logic and other
programming constructs into the configuration file reader.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan
8 error correction? I ask this questions, 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.
t, then you need to use multiplexing to
avoid having to have one channel per customer, which is excessive for
residential 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 - Huntsvill
n a virtual
machine will always means that you are subject 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
Che
> works but I am nervous to put 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 Driv
And it's not an error, so there's no need to do anything about it.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | j
coming media from the channel so
it can be discarded... 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 Da
digits
to select an extension to jump to.
Essentially, the only things that make sense to use in the 'h' extension
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. | Direc
eds 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
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsvil
s 'Asterisk', or even 'Asterisk 1.4', that
could 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. | D
maybe even a full second) before reporting 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 w
", &x)
> == 0))) ??
No. You aren't understanding the code :-) It's comparing a string buffer
against various patterns, and the string can't match all the patterns at
the same time.
This code is execut
.) in front of Asterisk, having 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
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
> wrote:
>> On 05/22/2010 02:07 AM, Deepesh D wrote:
>>
>>> I tried removing the dbhost and dbport entries and restarting asterisk.
>
fig: PostgreSQL
> RealTime: 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 Asteri
to use
> the unix socket for db connection?
You've specified *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 sp
do not know why Asterisk
would respond with 'recvonly', it should only do that when it thinks 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?
--
Ke
irectmedia'.
Asterisk 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 t
here's not really a
practical way to easily move TIFF/PDF files on and off of it, so 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 Dri
?
> This one has kept me up for 2 days now - if I had any hair i would be
> pulling it out now.
Like I said, 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
lable (both open source and commercial)
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.di
d.
> What CODECS 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 Dr
8FaxUdpEC:t38UDPFEC
>
> SIP/2.0 400 Bad Request
> ...
> CSeq: 102 INVITE
> Error-Info: ;cause="[line 023] SIP syntax error"
> Content-Length: 0
Which line is 'line 23' of the T.38 re-INVITE?
--
Kevin P. Fleming
Digium, Inc. | Director of Software Techno
such a way to avoid this requirement, but that is not possible 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 -
new version of Fax For
Asterisk that can take 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
Chec
r to
SendFAX() being started; this really should not be happening, as the
other endpoint should not re-INVITE until it knows that a FAX endpoint
is calling, but some of them do this anyway.
There is a fix for this problem in SVN Asterisk trunk already, and it
will be merged into the 1.6.2 branc
That means that Asterisk has to be able to understand the SDP content
that arrives so it can 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:
latest release from
the 1.6.2 branch, you'll probably have to wait until the next release 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 Sk
table schedule, which isn't possible when the virtualization
system can't guarantee that itself.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpflem
a nn-virtual machine 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 kern
d it was
primarily used for trunking between large 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
These should be pretty easy to find 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 a
your customer's restriction
that you can't change Asterisk versions.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@d
ing you like; we've had
tests 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...@dig
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
hing - the people 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
he behavior, we've found the culprit, and you can open an issue on
issues.asterisk.org so this can be investigated.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35
g mechanisms into Asterisk itself,
there is no other 'choke point' that can be employed. The decision was
made to do it in the RPMs because it was convenient to do so. If there's
no way to get the RPM tool to display any additional information about
why the conflict is present, that
it places 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 Technologie
be expected to work in Asterisk
1.4, there are a large number of API changes between those versions.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming |
ant to disable it, you'd have to modify the source code.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digiu
license agreements).
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 Softw
eatures.conf.
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 - Hunts
xactly that; general settings for the module, 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
licit template, 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.
--
AEL macro() is implemented using Gosub(), so you can use it as a
direct replacement. This is listed in the CHANGES file.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabbe
t contains the
changes required on its 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 | ja
e
configured them for minimally-sized port ranges based 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. | Direc
setting 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.c
ports in rtp.conf 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
D
it would 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 ou
S.
Right, computers don't have '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 calc
;m actually using AGI
>> script.
This sort 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
Chec
. 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. F
y. In this case, it is redirecting B's media back to itself in case
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: kpflemin
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. F
nvironment, where the media never goes through a switch
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 |
roblems on calls,
including calls 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.di
her options
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 | jab
e of how to
> access all 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
sk
, in other words Asterisk
> doesn't strip 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: kpf
to be bursty, so 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 :-)
--
27;send this frame _x_ times',
but that certainly could 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
y the behavior of most service 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.
sterisk/exts/gvoice.exten.conf
The #include is being processed before the [globals] is seen in
extensions.conf, so the [globals](+) in gvoice.exten.conf is the first
time the parser sees 'globals'. The simple fix for this is to put an
empty [globals] at the very top of extensions.conf, then in any
#x27;global' context or any other context.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Che
ll be sent to extension '100'. Asterisk refuses to
call that device again because it's already been called in that
particular 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 whe
mit the update 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 Technolog
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