, this will be
a major obstacle.
--
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.asterisk.org
, and it
is supported using SpanDSP and res_fax_spandsp. It is not yet supported
by Digium's Fax for Asterisk commercial FAX module.
--
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
it and the system you have it
installed in. This is a hardware issue, and should be pursed with Digium
Support.
--
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
the request could be associated with a peer 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 NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
as well.
--
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.asterisk.org
that is related, but it's still a bug :-) Unfortunately you've
reported this against an Asterisk 1.4.x release, which is in security
fix only mode, so even though it's a bug, there won't be a new 1.4.x
release available with a fix for it.
--
Kevin P. Fleming
Digium, Inc. | Director of Software
??
Did you look at the Request-URIs specified in the INVITE lines at the
beginning of the SIP messages? One specifies 's' as the target, the
other specifies '3292000101'.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com
On 12/27/2011 01:51 PM, Jonas Kellens wrote:
On 12/27/2011 08:45 PM, Kevin P. Fleming wrote:
On 12/27/2011 01:43 PM, Jonas Kellens wrote:
Hello list,
any idea why this call goes to the extension 3292000101 :
/INVITE sip:3292000...@ip.ip.ip.ip:5060 SIP/2.0
Call-ID: otrc74rls5c2pbyulb3hsjz
back into RTP packets on the way out. There is some
cost associated with this, but unless you are running a system that is
right on the edge of falling over due to channel load, it should be
tolerable.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem
-capable endpoint, you
can try that.
--
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.asterisk.org
by whatever is doing the SNAT/DNAT stuff.
--
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.asterisk.org
, and there's no
reason whatsoever for Asterisk to be reading it.
What is the GUI? There are lots of GUIs for Asterisk.
--
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
are free to do). It seems quite unlikely that the
presence of an 'a=rtpmap' line in the SDP for G.729 is going to cause a
call to have any problems.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan
likely won't ever see
it in practice :-)
--
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.asterisk.org
for such an option (which is why it was removed in the
Asterisk 1.6.x timeframe).
--
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
' that understands SIP and can fix up this
situation (and of course many Cisco phone users have Cisco routers that
do exactly this).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW
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.asterisk.org
On 12/07/2011 06:15 AM, Vieri wrote:
I can't upgrade this server to Dahdi and latest asterisk version...
In any case, according to the libss7 README, it should work with my software
versions.
What makes you think that? There is no support for SS7 in Asterisk 1.4.
--
Kevin P. Fleming
Digium
that the telco hasn't actually
'turned up' the span yet, because they don't usually do that until you
have your equipment plugged in and you call them to tell them that you
are ready for the span to be turned up.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem
green, then
at least your cabling/wiring are OK.
--
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.asterisk.org
. It is impossible to recover the password that was used
during the calculation of the response value (although given enough time
and CPU resources, it is possible go through a massive list of
possibilities and try each one until you find one that matches).
--
Kevin P. Fleming
Digium, Inc. | Director
, the 'call' signaling
still follows the same path it did originally, but the media stream path
can be shortened if the two endpoints are able to exchange media directly.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com
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.asterisk.org
a system like this before?
Here is one way:
https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+AIS
There are other methods documented on the wiki as well.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem
that extension(dont worry about this extension)?
No Asterisk does not support multiple registrations to the same SIP
account (AoR), but that is irrelevant in this case, because
registrations are not used for placing calls *to* Asterisk, only
receiving calls *from* Asterisk.
--
Kevin P
put CLI commands into cli.conf
and they will be run automatically when Asterisk starts. There are even
examples of doing this for 'sip set debug' in cli.conf.sample :-)
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com
On 11/08/2011 07:54 PM, Raj Mathur (राज माथुर) wrote:
On Wednesday 09 Nov 2011, Kevin P. Fleming wrote:
[snip]
* The GPLv2 places no restrictions on what you can 'write', it only
places restrictions on your distribution of things that you write
that could be considered 'derivative works
, anyone who has plans to
distribute Asterisk-derived works and wishes to do us under any license
other than the GPLv2 would be wise to consult legal counsel in their
area to learn how the license affects their plans.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber
-licensed software is illegal in your country? We don't even know
what country you live in, and even if we did, the answer to that
question is something you need to obtain from people who clearly
understand your country's laws.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
difference), you can certainly modify the module loader to skip this
check (you do have the source code, after all).
--
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
for a topic or
settings? Any example?
The process creates an index.html file under the doc/ subdirectory in
the Asterisk source tree; open it with a browser. However, you won't
find 'settings' in it at all; it will be documentation of Asterisk's
internal API calls for C development.
--
Kevin P
, that command was not documented to produce the
database results ordered in any particular order, so this change isn't a
bug, just a side-effect.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan
or for a work around.
... and that's why they haven't appeared yet. Very few people care about
them at this point, because web-browser based monitoring of large
numbers of extensions tends to be much more efficient and easier to use
than large button panels on a phone.
--
Kevin P. Fleming
Digium
be possible for directmedia to be enabled
for RTP and not interfere with UDPTL, but there could still be lingering
problems there.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW
will not be
mixed into the conference), then Asterisk *could* send it a message
telling it to not bother sending any audio. That does not happen right
now, but wouldn't be a terribly difficult patch to write.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem
On 10/11/2011 02:04 PM, Jeremy Kister wrote:
On 10/11/2011 11:48 AM, Kevin P. Fleming wrote:
Well, as a starting point, I'd suggest disabling directmedia
(canreinvite) on s3. It should be possible for directmedia to be enabled
for RTP and not interfere with UDPTL, but there could still
features that have been recently added to chan_misdn,
you'll need to use the branches of mISDN at
http://svn.digium.com/svn/thirdparty
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis
On 10/08/2011 05:17 AM, Steve Underwood wrote:
On 10/08/2011 02:50 AM, Kevin P. Fleming wrote:
On 10/07/2011 07:46 AM, Administrator TOOTAI wrote:
Hi,
I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken
from deb http://packages.asterisk.org/deb lucid main) including dahdi
On 10/08/2011 05:21 AM, Steve Underwood wrote:
On 10/08/2011 04:04 AM, Kevin P. Fleming wrote:
On 10/07/2011 02:20 PM, James Sharp wrote:
On 10/07/2011 12:27 AM, Nasir Iqbal wrote:
Check firewall and NAT settings!
Monitoring sip and media flow from asterisk cli can help, use sip set
debug
specify the 'c' option?
--
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.asterisk.org
the UDPTL packets themselves
so we can see what they contained.
--
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
with extension not application.
No, none of that is relevant. It's perfectly acceptable to call
SendFAX() on a CLI/AMI/spool-originated channel.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan
On 10/07/2011 03:29 PM, James Sharp wrote:
On 10/07/2011 04:04 PM, Kevin P. Fleming wrote:
First, we can see that Gafachi's T.38 implementation still has some
breakage in it (although the problems are ones that Asterisk has been
taught to deal with). In its 200 OK to the T.38 re-INVITE, it has
format like GSM, G.729 or something else supported in Asterisk.
--
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
in.
--
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.asterisk.org
output.
--
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.asterisk.org
On 09/30/2011 07:49 AM, Administrator TOOTAI wrote:
Le 30/09/2011 14:05, Kevin P. Fleming a écrit :
On 09/30/2011 03:56 AM, Administrator TOOTAI wrote:
Hi list,
we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is
Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS
), but not the other way around to my knowledge.
Once an HDLC network link has been setup in the kernel's HDLC layer, I
don't believe it can be shrunk or grown.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan
asterisk v1.8 chan_misdn works only
with linux kernelv2.6.24 which is quite old.
chan_misdn only supports mISDN version 1.
--
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
is correct, as of DAHDI 2.4 and later.
--
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.asterisk.org
(Illegal Instruction) so my script might be seg faulting
somewhere? Should I be going after this? It is a php script and php doesn't log
anything for these instances.
No, 4 isn't SIGILL; result codes generated by uncaptured signals are
always negative, I believe.
--
Kevin P. Fleming
Digium
configuration (using the faxbuffers option in
chan_dahdi.conf, for example), such a system can be setup to work very,
very close to 100% of the time.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan
an insurance policy against timing slips.
--
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.asterisk.org
without help (which is where
T.38 and V.150 enter the picture).
--
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
or Digium's Fax for
Asterisk backends.
--
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.asterisk.org
channel could be looped back towards its source, on demand, with
nearly zero latency.
--
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
port) has answered or not, it assumes the
outgoing call is 'answered' as soon as dialing has been completed.
Because of this, the calling channel is bridged to the called channel as
soon as dialing has been completed, and the calling party will hear the
progress of the outbound call.
--
Kevin P
, if not 128ms.
--
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.asterisk.org
will be able to call.
chan_sip does not support specification of the password to be used for
authentication in the dial string itself; your :password suffix is
just being sent to the target system and it is trying to find a matching
extension in the dialplan (and failing).
--
Kevin P. Fleming
window, you'll have to find another way of generating echo for it to be
tested against.
--
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
exactly what you are looking for; I suggest you look him up and find out
what state it is in, and see whether you can help test it.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW
*any*
network element that packetizes the audio will result in a delay longer
than 16ms.
--
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
, but I have no idea whether it does 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 NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
On 09/08/2011 10:04 AM, Daniel Tryba wrote:
On Thu, Sep 08, 2011 at 08:38:39AM -0500, Kevin P. Fleming wrote:
The following just works for any SIP client (without
overlap dialing):
exten = _X.,1,Answer()
exten = _X.,n,Dial(${TRUNK})
Unless I'm mis-remembering, this was the point of adding
On 09/05/2011 03:05 AM, Hans Witvliet wrote:
On Fri, 2011-09-02 at 11:33 -0500, Kevin P. Fleming wrote:
On 09/01/2011 04:39 PM, Hans Witvliet wrote:
From the asterisk-bible and the wiki's i learned that it is possible to
let asterisk do some of the presense-info by means of the jabber.conf
-config - please install and try again
This is a bug in the configure script, but in the meantime, you should
be able to use --without-pwlib to avoid it, as long as you aren't
trying to build chan_h323.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem
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.asterisk.org
to distribute the information between
the servers.
--
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.asterisk.org
developed after the first 1.8 release was made.
--
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.asterisk.org
data..., look for valid ascii, and otherwise
put out TDM modem tones with no data content for 1 second and then pick up
the data as it catches up.
So you want to develop the equivalent of T.38 for dial up?
It already exists; it's called V.150.
--
Kevin P. Fleming
Digium, Inc. | Director
provide all the information you need.
The sip debug logs I can post here but I need to change the real IPs,
which is easy to do because it will be a text file.
I appreciate your time and effort in helping us find the roout cause.
Yes, that is the correct location.
--
Kevin P. Fleming
Digium, Inc
with signaling and media.
PLease let me know what other thing you need you need me to provide.
I've already started looking at the packet capture; I'll follow up on
the issue itself (ASTERISK-18394 for those following along at home).
--
Kevin P. Fleming
Digium, Inc. | Director of Software
just put random content in a SIP request
or response message body; the message body is usually of a defined type
(application/sdp, for example), and has rules about what it can and
cannot contain.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com
verbose/debug levels. Is there a way you can produce that and provide it
to us without having to reveal confidential information? If not, we can
create a private issue on the issue tracker for you to have a place to
upload your files without them being visible to the public.
--
Kevin P. Fleming
as if the endpoint had placed the call
itself in any of the SIP discussion lists I frequent... so I'm pretty
sure there's no standard way to do this.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445
are maintaining them.
--
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.asterisk.org
(in chan_dahdi) does have some
call progress detection support which you can enable, but it may be
anywhere from completely useless to only partially reliable for you,
depending on your specific situation (country, indication tone patterns,
provider, line quality, etc.).
--
Kevin P. Fleming
-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...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW
it won't be forgotten. Thanks.
--
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.asterisk.org
be:
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
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.
--
Kevin P. Fleming
Digium, Inc. | Director of Software
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
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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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
Digium, Inc. | Director
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 of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem
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 Drive
-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: kpfleming
445 Jan
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: kpflem
subject :-)
--
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.asterisk.org
of 'Logoff' and
Asterisk is complaining.
Well, he's sending DBPut before reading the result of Login as well.
--
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
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
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
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.asterisk.org
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 us out
, 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.com | Skype
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 - USA
Check us out at www.digium.com
) to find out if those features can be disabled.
--
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.asterisk.org
).
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
Jabber
in
that situation.
--
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.asterisk.org
of 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.asterisk.org
on it, they'll
attack it.
--
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.asterisk.org
.
--
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.asterisk.org
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