.
This could be caused by a number of reasons, but the most likely is that
your syntax isn't correct above. Try either:
channel originate sip/iptel-out/echo Application playback vm/net_ring
or
channel originate sip/e...@iptel-out Application playback vm/net_ring
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if there is any background noise at all. If
this is documented, point me to where and I'll gladly do my reading.
You can adjust them manually with the txgain= and rxgain= settings in
chan_dahdi.conf.
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On Mon, 2010-05-31 at 22:08 +0200, Jonas Kellens wrote:
Is there yet a seperator that actually works to define multiple mail
addresses ?
Not that I'm aware of. I simply create an alias on the mail server that
then forwards to all the recipients.
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, Asterisk would
automagically create a channel variable named USERID with a value of
jsmith every time this device made a call into Asterisk.
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On Tue, 2010-05-25 at 12:07 -0600, Steve Johnson wrote:
How can you determine how many are already in the conference bridge?
I don't know that there's a way to do it automagically within
ConfBridge. I use the GROUP() and GROUP_COUNT() functions to do these
sorts of things.
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?
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get overwritten.
It's certainly worth a shot...
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:
;4200 = 9855,Mark Spencer,marks...@linux-support.net,
mypa...@digium.com,attach=no|serveremail=mya...@digium.com|tz=central|
maxmsg=10
See how we set this particular mailbox to only have a maximum of ten
messages?
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. While it's not
verb noun, most (if not all) of the commands in the Asterisk CLI
should follow the module verb noun model.
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://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
That's not correct. DIALSTATUS will be set whether or not you've got
qualify=yes in the peer definition.
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, and be inherited by the spawned call. Am I missing something
obvious?
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asterisk
On Tue, 2010-04-13 at 13:59 -0500, Danny Nicholas wrote:
They actually do have a timestamp, in a manner of speaking. The uniqueid
field is a pseudo-unixtime stamp.
While correct, it's a timestamp of when the call *started*, not when the
event happened.
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(/usr/lib). I had no problem compiling cdr_odbc on my test
server(CentOS 4.6), however following the same steps on my production server
(CentOS 5.4) gives no joy.
Install the 'libtool-ltdl' and 'libtool-ltdl-devel' packages, and then re-run
./configure.
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of Asterisk, you could use
the res_jabber and the JABBER_STATUS function to see if they're marked
as available in their XMPP IM client. (Most IM clients will set the
status to away when the screensaver kicks in.)
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well enough in my tests to warrant
its use.)
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will then only look in the switch if it doesn't find a match in
extensions.conf.
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, or otherwise ensure that Asterisk doesn't play
one greeting for callers with one codec and another greeting for callers
using another codec.
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turning on automon.
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this out there for discussion (and hopefully more
enlightenment).
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On Sun, 2009-12-06 at 08:49 -0500, Dan Journo wrote:
I’m trying to figure out how to limit the number of concurrent calls a
client can make.
I prefer to use the GROUP() and GROUP_COUNT() dialplan functions to
enforce arbitrary call limits in Asterisk
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have corrected them by
now.
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outside of my control.
Type core show channels at the Asterisk CLI to see each channel, and
what it's being bridged to.
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. Is there a particular reason you want to pull *all*
of them?
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Asterisk and your VoIP provider.
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On Thu, 2009-11-12 at 08:53 -0600, Cary Fitch wrote:
Digital 64K telco sounds very good as a phone conversation.
Digital 64k audio coming across a T1 is essentially identical to the
ulaw codec in VoIP. Digital 64k audio coming across an E1 is
essentially identical to the alaw codec.
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problem.
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in Asterisk 1.4
and/or counteronpeer=yes in Asterisk 1.6.0 and later.
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at the same time. Extension 123456 modifies the CallerID and
then calls Charlie's cell phone number.
I realize that chan_local takes a bit of work to understand, but trust
me -- once you get used to it, you'll wonder how you got along without
it.
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), and there's enough
interest in the area that we might start up a local Asterisk users group
in the area. What part of Virginia are you from?
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syntax for the Voicemail() dialplan application.
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command ? Will asterisk
stack commands or will it stop the first one to execute the second one
?
If you want non-blocking (asynchronous) commands, check out the
ExternalIVR interface instead of using AGI.
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for that brand of phone.
I do this with my polycom phones and it works well. Don't know if it
works with other brands of phones.
I've done this on a number of different phones, using both the ISC dhcpd
server as well as dnsmasq. I've never encountered any problems with it.
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wrong here... try mailbox=...@a106...@a10
instead. Contrary to what others on this thread might lead you to
believe, this should actually work. :-)
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can identify the responses with the
corresponding action based on the ActionID.
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you can guarantee that it's going to be
unique across concurrent calls. Otherwise, it's not likely to be very
useful to you in the long run.
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- Danny Nicholas da...@debsinc.com wrote:
Two questions: 1. do you need an ActionID line?
Danny,
It's *always* considered best practice to have an ActionID line in AMI
commands, so that you can easily differentiate the responses, especially to
asynchronous commands.
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remember if it's available on the 1.6.1 branch. I know it's not
available on the 1.6.0 branch.)
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without having any experience on
Polycom IP phones and Digium cards, as long as you know how to use
Asterisk itself.
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message?
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asterisk-users mailing list
://issues.asterisk.org/
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cell phone use.
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be very
straightforward to pass the practical portion of the exam. If you're an
Asterisk novice, you probably won't pass (even if you do copy/paste
configs from a website).
If you have further questions about the dCAP exam, I'd be happy to do
what I can to answer them.
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, the general idea is this: A small company has hired you to build
a typical small-business PBX using Asterisk, and you have 90 minutes to
get it up and running. Given the time constraint, we really stick to
the basics, so there shouldn't be anything unexpected during the test.
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.
That being said, we absolutely support *hangup* supervision.)
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stanza
for each day you want to match on (Tuesday, Thursday), etc. unless
they're in a range (tue-thu, for example).
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like func_odbc), or using the Asterisk Manager Interface to
poll for the data.
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On Wed, 2009-08-05 at 13:12 -0500, Jon Moore wrote:
I have in my sip.conf the following
[jon.moore]
type=friend
mailbox=8100,8150
In voicemail.conf, both mailboxes are defined.
Have you tried 81008150 (using an ampersand instead of a comma)?
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of another extension, or to the status
of a voicemail box.
A registration is where one SIP device tells another Hey, I'm over
here. If you get any calls for me, send them to me at this IP address
and port.
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On Tue, 2009-08-04 at 09:45 -0500, Danny Nicholas wrote:
This is a hack solution;
There's nothing hackish about it. It's a very useful tool for
shortening the call path and freeing up bearer channels that would
otherwise be tied up in bridging the calls.
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... Asterisk doesn't currently do anything with the
facility message coming back from the telco when the call ends.
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options that might have changed, etc.
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asterisk
a more
expensive card, but hopefully you have a better understanding of why we
don't use modems as FXO devices. If your time and sanity are worth
anything at all, it's a worthwhile investment to buy a good solid
telephony card.
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xxx.xxx.xxx.0/255.255.255.0, which probably
isn't what you want.
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the
parser understand them both?
That was one of my pet peeves with Asterisk 1.4 and earlier. I asked
the developers to address it, and it's my understanding that in 1.6.x
and later that Asterisk will accept the word signaling with either
one, two, or even three 'l's. :-)
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or transcode to/from G.729.
3) If I use G.729 for voice communications and GSM for voice mail
sounds, does Asterisk execute trascoding ???
It will, if you have added the G.729 codec.
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the
extension state changes, Asterisk will send a SIP NOTIFY to the phone to
let it know that the subscribed hint has changed states.
I know you're only trying to help, but please don't muddy the water by
telling people that MWI and BLFs are the same thing.
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this out without asking
here, but it's been 2 weeks and I'm still failing.
Have you tried mailbox=...@default? It appears as though you need to
specify a voicemail context.
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download, check out
www.asteriskdocs.org.
There are obviously many other ways to do it.
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is that it should work for DTMF
and flash-based transfers. I'm a little less sure about SIP-initiated
transfers.)
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://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect111_tt1363.html.
They may be a bit out of date (as the Asterisk GUI has changed quite a bit
since we wrote the book), but it should help you get started.
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.
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On Tue, 2009-07-07 at 10:47 -0400, Jeremy Winder wrote:
It seemed to me cron was going to be the best solution.
Sounds like overkill to me... why not just use a GotoIfTime clause in
your dialplan?
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of Asterisk (1.6.0 and later, if I
recall) support SMDI.
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Status
Exten: 555
Context: lab
Hint: SIP/linksys
Status: 0
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the
phones and gateways. All you'd need to worry about would be licenses
for the G.729 transcoding that Asterisk is doing.
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for more
details.
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investigate further?
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before, but I'd be happy to go over it again if anyone wants
me to.
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or you can obviously buy a dead-tree version of the book from you
favorite bookseller.
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the second digit, option
6 gets sorted above option 7 because it is more constrained. After the
third digit, however, option 6 is eliminated because the last digit
can't be a zero. That means that Option 7 will match.
Clear as mud?
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for
more information.
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to doing more with it in the next few months.
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let you communicate via IMAP,
right?)
In short, there are a lot of exciting things happening in the world of
Asterisk with regards to unified communications.
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benefit of supporting messages
in a variety of languages.)
Anyway, just my 2 cents (before taxes)...
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, not for the messages themselves, so
obviously my use of the CHANNEL function to set the language was
short-sighted.
Thanks for keeping me honest and on my toes!
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of
Asterisk. If I remember correctly, Asterisk 1.6.0 and later use the DNS
Manager (see dnsmgr.conf) to periodically re-resolve DNS names.
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to remove any spaces from around the question mark
(after your expression).
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is.
From there, you can start to track down the source of the problem one
network segment at a time. For example... is the poor audio being
caused by network problems between the phone and Asterisk, or between
Asterisk and your upstream provider.
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and effort improving
the performance of the internal structures between the 1.4 branch and
the 1.6.0 branch... if I were you, I'd at least give the 1.6.0 branch a
shot.
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. Why not just
set an absolute timeout on the channels? Something like:
exten = 123,1,Set(TIMEOUT(absolute)=3600)
exten = 123,n,MeetMe(blah,d)
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that have been applied since the
1.6.0.9 release.
Does that make sense?
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*
experimental, and in my experience causes more problems than it's worth.
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channel dies at 6:00 and John's dies at 6:01. (You could obviously add
dialplan logic to calculate a smaller timeout value for John's call, but
I'll leave that as an exercise for the reader.)
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remember which off the top of
my head), at least *one* of the calls must be inbound from the telco to
your Asterisk box.
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.)
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has no
way of telling whether or not the remote party has answered the call or
not. This is entirely due to the way analog signaling works, and works
exactly the same under both Zaptel and DAHDI.
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in there?
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PlayTones() to decrease the volume on the Tx side, and
then possibly restore it after calling StopPlayTones().
I haven't tested it to see if it works.
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-retransmit.txt.
Did you read doc/sip-retransmit.txt? As it explains there, the remote
device didn't respond to our critical SIP packet, so Asterisk had no
other choice but to terminate the call. You need to figure out why the
SIP responses aren't getting back to Asterisk.
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the output of core show channels to see what
application the hung channels are in? I'd start there.
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anyone know how to distribute the licenses among
several servers?
Please open a support ticket with Digium's support department... they'll
take care of your problem for you.
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we can add yet another option.)
It seems silly to have to recompile just to get this functionality.
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)
exten= 1246463,n,Wait(5)
exten= 1246463,n,Hangup
exten= 6463,1,Dial(SIP/8003,60,rT)
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right -- the CDR information is for the entire call.
Instead, look at the queue log (typically written to
/var/log/asterisk/queue_log). It will tell you most (if not all) of the
information you need for creating call queue reports.
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Training Manager
Digium, Inc
if this would update the CDR records for the original call!)
I hope that's enough documentation to get you started! Please let us
know how it works out for you!
--
Jared Smith
Training Manager
Digium, Inc.
___
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to
using the System() application and finding a way to send your email from
the system command line. Not impossible by any stretch of the
imagination... it just takes a bit more work.
--
Jared Smith
Training Manager
Digium, Inc.
___
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. Is Asterisk running? If
so, can you find the asterisk.ctl file that was created when it was
started?
--
Jared Smith
Training Manager
Digium, Inc.
___
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