usually it's better to ask about the specific
features you are interested 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 o
to see what it will bring. I'll be
forever thankful for the opportunity that Digium and the Asterisk
community provided me to learn, grow and find the place where my skills
and experience are the most valuable (to both myself and my employer).
--
Kevin P. Fleming
Digium, Inc. | Director o
arly biased, but I suggest you consider the Digium G100/G200
gateways. We've had really good feedback from users since they were
released, and they certainly meet all of your needs listed above (and
there are a few more, like T.38 FAX support on all channels).
--
Kevin P. Fleming
Digium,
te 'conventional
wisdom' from the world!
--
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
ou disable echo cancellers? That's a terrible idea.
--
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 &am
phones are the statuses that
Asterisk itself generates based on the phones' activity.
--
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 o
e recording the call without overloading its CPU.
--
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.co
both legs of the call are using the same codec, then normally
Asterisk would not modify the audio in any way at 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
way code (since it has been
enabled) will realize that the DAHDI channel is not T.38 capable, and it
will step in to provide T.38 gateway services.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445
g standard email libraries in many scripting languages.
--
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 &a
ou to do), the email client that receives
the HTML is going to treat it as plain text, which is what you saw.
--
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
much help there. I know that Audacity will open them and play the
audio properly, because that's what we used when we developed this
'audio capture' feature.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com
to ask the person(s) who made the channel driver
you are using, since it's not part of Asterisk itself.
--
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
ies of the main Asterisk log
directory.
--
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
ew: V.34 not supported, will be ignored." Is v34 only supported with
SpanDSP?
Those docs are in error. V.34 is not supported. I'll notify our
documentation people. Thanks for the report.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.co
ers, result in
interoperability problems.
--
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
ch would be overkill.
Benny, are you aware of some other method to accomplish 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 - Huntsville, AL 35806 - USA
Check us o
es, this is the situation I referred to earlier. In your case, it's all
on one interface, but the server has multiple addresses on the *same*
network, and thus it cannot know (without help) with address should be
used for outbound packets.
--
Kevin P. Fleming
Digium, Inc. | Director of Software
this
simple in order to cause the IP stack on the Asterisk server to choose
the wrong source IP address for outbound packets.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Dri
On 07/12/2012 09:19 AM, Benny Amorsen wrote:
"Kevin P. Fleming" writes:
That's quite interesting; can you describe a scenario where this occurs?
Imagine you have a server with two interfaces, eth0 with 192.168.1.1/24
and eth1 with 10.0.2.1/24. Further imagine that you wis
On 07/11/2012 07:51 AM, Olle E. Johansson wrote:
10 jul 2012 kl. 20:50 skrev Kevin P. Fleming:
On 07/10/2012 03:24 AM, Olle E. Johansson wrote:
The Asterisk SIP channel has no knowledge about interfaces and can't
bind to a specific interface for communication. In fact, it's a well
On 07/10/2012 01:50 PM, Carlos Alvarez wrote:
On Tue, Jul 10, 2012 at 11:46 AM, Kevin P. Fleming mailto:kpflem...@digium.com>> wrote:
This can be done using Digium phones; they have built-in support for
selecting which 'user' they should be when they are reconfigured.
g source IP address,
since it does not specify the source IP address at all. If this is
occurring, it must involve the operating system's IP stack in some fashion.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | S
iguration state and wait for
someone tell it which extension it should 'be'... when the user returns
home, they can 'steal' the extension back from the office.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@
hough it's pretty
uncommon for users to have 'turn on' their phones.
--
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
y to determine whether
the devices you list as members are currently connected to Asterisk 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 358
whether or not Asterisk will respond appropriately to a
re-INVITE received *from* a SIP endpoint (to which Asterisk should
always respond properly, unless the re-INVITE is malformed in some way
or is unacceptable).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@d
On 06/27/2012 09:30 PM, David Cunningham wrote:
Would anyone else know if Asterisk allows use of SpanDSP's time zone
conversion?
No, SendFAX (in res_fax) doesn't currently offer the ability to do what
you are asking about.
--
Kevin P. Fleming
Digium, Inc. | Director o
damage to users' ears), and the Digium phones are no exception.
If you are finding that the volume produced by common SIP phones is too
low and you can't make it loud enough, I'd bet that the problem is not
in the phones, but in your environment or your ears :-)
--
Kevin P. Flem
secure to VPN.
SIP over TLS (what used to be called SSL) is what secures the SIP
signaling. SRTP is for securing media streams.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davi
it refused to accept a re-INVITE from Asterisk that
wanted to switch the SIP channel to T.38 mode.
--
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
ep-by-step instructions on setting up an FXS port for use with an
analog telephone.
--
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 o
is my next step once I have the analog lines
working).
Have you read any of the O'Reilly Asterisk books? They will help you
learn quite a lot about Asterisk, and they are available online.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP:
said that when
the audio is directed at endpoints that have a proper jitter buffer,
there is no issue. If you send the call over SIP to this 'SV8300' device
and still have audio issues, that would imply that this device does not
have a jitter buffer capable of handling this level of
available to you or not.
Does anybody have any suggestions here?
It sounds like the lack of a proper jitter buffer (of adequate size) is
the issue here, since when the audio is directed at endpoints outside of
Asterisk that have them, the audio is as you'd expect it to be.
--
Kevin P.
On 06/14/2012 05:23 PM, asterisk users wrote:
Is there a detailed application note in the Digium wiki (or anywhere
else for that matter) about these implementing features under
Asterisk/Switchvox?
Not yet, I don't believe.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technol
or us to consider
moving our organization from Mitel to Digium/Asterisk/Switchvox.
How much of this can be done with Asterisk/Switchvox and, say, the
Digium D70 phone with dynamic button display?
Most of it, I think. Give them a try!
--
Kevin P. Fleming
Digium, Inc. | Director of Software
On 06/14/2012 04:20 AM, [Digital^Dude] ® wrote:
How can I set a hard limit to the number of Local channels asterisk can
spawn?
chan_local does not have a mechanism to do this.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem
Thanks in Advance!!
No, there is not any way to do 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
oose, they will all produce
identical signal levels ('voice volume') when plugged into your
telephony circuit(s).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - H
s for doing this. I have used
VMAuthenticate but I would like more flexibility than what this offers
What do you need that VMAuthenticate does not offer?
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpflemin
On 06/06/2012 09:46 AM, Eric Wieling wrote:
For some reason 1.4.4.x was not reading chan_dahdi.conf. When I symlinked it
to zapata.conf it worked.
That means Asterisk (1.4.4.x?) was built against Zaptel, not DAHDI.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber
that would be greatly appreciated.
The Hx8 User's Manual (here:
http://docs.digium.com/H8/hx8_series_manual.pdf) has an entire chapter
on software installation and configuration, including DAHDI, libpri and
Asterisk.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
J
ase 'should'
in RFC 3261.
--
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 a
On 05/25/2012 06:30 PM, Dave George wrote:
How can I enable the option to allow asterisk to maintain the RTP stream
during DTMF collection?
If it's the problem I hypothesized it was, you can set
'transmit_silence=yes' in your asterisk.conf file.
--
Kevin P. Fleming
Digium,
be represented
in CDRs.
--
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 &am
x27;t available in your price range.
--
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
(since 2.4, I think)
has made this information available in /proc/dahdi.
--
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
was running and waiting for input from the
caller; if your version is older than this, then that could explain what
you are seeing. That's just a mildly-educated guess though.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem..
should fit just fine into PCI-X
slots. Do you mean PCI-Express instead? That's very different.
--
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
Che
the benefits of his activities across the community!
--
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
are
just passing T.38 *through* Asterisk, between an ATA and the AudioCodes
gateway. In that case, 'updtl debug' on the Asterisk CLI will show you
the UDPTL traffic flowing through Asterisk.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digi
see in the logs if T.38 packets sending and
see somehow its debugs? Or I should just be better off with capturing
sip data through tcpdump?
This will depend on what you are asking the Asterisk 10 system to *do*
with T.38. Are you sending FAXes from it, or receiving FAXes into it, or
something else
is of utmost relevance:
http://hylafax.sourceforge.net/docs/fax-over-voip.pdf
Absolutely correct.
--
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
since we don't know what version
of Asterisk you are using. However, in Asterisk 10, there is a
channel-agnostic FAX detection function that can be applied to any
channel type, so at a minimum that is one way to solve your problem.
--
Kevin P. Fleming
Digium, Inc. | Director of Software
xperiences a failure.
You'd be much better off to at least split the load across two machines,
both of which should be large enough to handle the entire load when
necessary.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpfle
am is going to
consume vastly more bandwidth than the audio stream.
--
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.digiu
ittle documentation.
There is no support in chan_dahdi to make Asterisk behave *as* a GR-303
channel bank.
--
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
Che
.
--
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
ng on the pbx side that we are hoping to transfer media to?
No.
3) How long into the call before the media is transferred over?
It should happen quite quickly after the call is answered.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP
both of your systems are 1.6.2.x or later, you can use
'directmedia' on all of 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
Chec
r of reasons that would impede native bridging (transcoding,
recording, etc.).
It seems like you have the configuration set up (mostly) properly, so in
order to know what is going on you're going to have to post a more
complete log snippet, including 'sip debug' output.
--
Kevin P.
we have to pay for
some license?
Your questions are answered on the Certified Asterisk page on the Digium
website:
http://www1.digium.com/en/products/asterisk/certified-asterisk
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem
at license and get another one. There is no 'buying'
licenses, they are free.
--
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
while tapping on the
table with another cell and being stunned with the magnitude of the
delay and that most people manage to carry on conversations without
noticing.
Yes, cellular networks have largish latencies, but no jitter.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Techno
; they are the *same* feature. If the document(s)
you read didn't make that clear, the authors did you a disservice.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsvi
useful.
--
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
tributions.
--
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
released. The D40 and D50 have the same size
screen, but it is smaller than the one on the D70. The D40 and D50 do
not have a hard 'Apps' button, but they do have an on-screen softkey for
access to applications.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jab
-series failover appliances will work with any device
that uses the appropriate type of PSTN circuits (digital or analog,
depending on the R-series model), even a legacy PBX.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com
ormance penalty of registrations.
However, it would not surprise me in the least if Asterisk 1.8.x and
later handled that volume of registrations without much of a problem.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@dig
of them) to the equation.
--
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 &am
OpenWRT, a SheevaPlug, or any number of
tiny, low power embedded devices. As Carlos already said, *any* decent
x86 box produced in the last five years would be able to handle this
without any noticeable CPU load at all.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
J
uot;shows
extension is rejected", but extensions don't get rejected. Extensions
can be 'not found', but that's very different from rejected.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com |
releases of DAHDI and Asterisk
should have support for 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
still move through solid materials :-)
--
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 a
uld not even be
sent to Asterisk at all (it should go to wherever the URI resolves to).
--
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
ectory of the source code tree.
--
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
building apps is usable? Can you build powerfull apps?
Examples?
The phone app SDK has not been released yet, it's still under development.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan
DAHDI,
libpri and Asterisk working together.
--
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
be tested by using the CONNECTEDLINE()
dialplan function to send anything desired to a phone that is in a call
with 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
og modules except the TDM2400P.
--
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
ing/spying,
music-on-hold, conferencing, etc.) Given that, what you really want is a
pure SIP proxy like Kamailio or OpenSIPs.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW -
e, because ring-back tones
vary greatly, and they might not even be traditional ring-back (many
mobile providers offer 'music ringback' to their subscribers).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com |
pdate, instead the target device is located when a call requests it.
There are many examples of this on the Internet... use your favorite
search engine.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
44
On 04/25/2012 05:29 PM, Eric Wieling wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Wednesday, April 25, 2012 6:25 PM
To: asterisk-users@lists.digium.com
Subject: Re
exist inside the
Queue.
or maybe an application which written from scratch... can be help full.
It's hard to parse what you are saying, but yes... it would be possible
for someone to write code to do what you want to do.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
J
( and there so louder ) on the channel! without
stoping MOH?
This is an interesting idea, but at this time Asterisk's app_queue has
no ability to do what you are asking for.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: k
On Apr 25, 2012, at 5:00 PM, "Kevin P. Fleming" wrote:
On 04/25/2012 07:08 AM, Bryant Zimmerman wrote:
I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
track the actual SIP response code as well. How do I get access to it
durring the hangup?
It's rather hard
on of Asterisk you are using. In some versions there is a SIP_CAUSE
feature that can be used to extract that information (although this has
been reimplemented for Asterisk 11 in a way that doesn't affect
performance as much as the old method did).
--
Kevin P. Fleming
Digium, Inc. | Director o
ed it. I don't think we'd want to merge patches
that added support for either of those mechanisms.
--
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 35
still use
the most recent release from that branch.
One final note: please don't reply to list posters' personal email
addresses unless they ask you to do so. The list is configured to force
replies to go back to the list, and that's done for a reason.
--
Kevin P. Fleming
Digiu
what it was able to find and
what it was not able to find, but the menuselect information is a
reasonable next step.
--
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
message to go
away, add 'noload => codec_dahdi' to your /etc/asterisk/modules.conf file.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Driv
mance.
--
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
ented' on the G.729 download selector page here:
http://www.digium.com/en/docs/G729/g729-download.php
Did you use that download selector, or go directly to the
downloads.digium.com site to grab the files?
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@
.
--
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
On 04/18/2012 11:23 AM, Dan Austin wrote:
Kevin P. Fleming wrote:
This is a valid point, and we'll get this corrected. Our package
repository should have packages for Asterisk 10, but it doesn't.
How likely is it that a Centos 6 repo might be setup at the same time?
It's o
rcial product and you are
entitled to technical support.
The simple answer to your question is no, there are no known
incompatibilities between Asterisk 1.8 and Digium's G.729 codec modules
(if there were, we'd fix them).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Techn
On 04/18/2012 06:08 AM, Niccolò Belli wrote:
Hi,
Il 18/04/2012 00:39, Kevin P. Fleming ha scritto:
You guys know that it works in Asterisk 10, but you say you can't use
Asterisk 10 for some reason that I don't understand.
1) No Debian packages for v10. If you have to maintain lots
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