ip dialog (long ID) INVITE.
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
s hard to hear 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.
-
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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Jared Smith
"setvar=USERID=jsmith" in a user/peer/friend definition, 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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Jared Smith
Sr. Trainer
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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.
-
functions in the dialplan to enforce call
limits.
Clear as mud?
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he
Request URI wouldn't get overwritten.
It's certainly worth a shot...
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; command was changed to "channel request hangup". 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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Jared Smith
Digium, Inc.
--
l.conf file that comes
with Asterisk:
;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 te
sip.conf or iax.conf).
> http://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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Jared Smith
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__
nherited by the next spawned channel, but go no further.
If you define a variable with two underscores (say, __TRANSFER_CONTEXT), then
it will get inherited by the next spawned channel, and any channels spawned by
that channel, and so forth. Obviously defining it without any underscores at
all mean
erisk, but Called Party ID will
be supported in Asterisk 1.8. If you're adventurous, you can try out
trunk now on a development machine and ensure that it's working the way
you want it to before Asterisk 1.8 is released.
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Jared Smith
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et set on the incoming call from that particular
user, and be inherited by the spawned call. Am I missing something
obvious?
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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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Jared Smith
ns (/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
./c
a relatively modern version 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
it worked well enough in my tests to warrant
its use.)
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h
a switch statement. Asterisk
will then only look in the switch if it doesn't find a match in
extensions.conf.
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asterisk-us
ts, 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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his is the case, but thought
I'd throw this out there for discussion (and hopefully more
enlightenment).
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e between the * and 1 keys
when turning on automon.
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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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ere any regressions of this nature in the transition
from Zaptel to DAHDI, rest assured that we would have corrected them by
now.
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> the call routing to determine if Asterisk is doing this or if it's
> occurring 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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arballs you have selected in
"make menuselect". Is there a particular reason you want to pull *all*
of them?
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To
tself
or between 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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are of your problem.
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erisk will dial the two SIP devices and extension
123456 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 a
also need to investigate "limitonpeer=yes" in Asterisk 1.4
and/or "counteronpeer=yes" in Asterisk 1.6.0 and later.
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ear Fredericksburg), 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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1...@default,u)
You can always type "core show application voicemail" at the Asterisk
CLI to see the complete syntax for the Voicemail() dialplan application.
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server as well as dnsmasq. I've never encountered any problems with it.
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ediately followed by another 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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Jared Smith
Tr
think you've got the syntax wrong here... try mailbox=...@a10&6...@a10
instead. Contrary to what others on this thread might lead you to
believe, this should actually work. :-)
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that means 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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at
same ActionID, so that you can identify the responses with the
corresponding action based on the ActionID.
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AstriCon 2009 - October
- "Danny Nicholas" 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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Jared Smit
nch, but I
don't 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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PSTN or
VoIP connectivity to do so.
In a nutshell, you can pass the test 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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ion to have it skip the introductory
message?
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n the exact details of the dCAP
exam, 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 unexpect
l capacity, it should 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 an
against cell phone use.
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asterisk-users mail
r at http://issues.asterisk.org/
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asterisk-users mailin
ut, you'll need a separate GotoIfTime 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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reversal to indicate far-end answer.
That being said, we absolutely support *hangup* supervision.)
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AstriCon 2009 - October 13
se (using
something like func_odbc), or using the Asterisk Manager Interface to
poll for the data.
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saying that the phone is
not registered.
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as
wants to subscribe to the state 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."
--
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 8100&8150 (using an ampersand instead of a comma)?
e point that the
transfer happens... Asterisk doesn't currently do anything with the
facility message coming back from the telco when the call ends.
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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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re details of
configuration options that might have changed, etc.
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Register N
Aculab.
I know that probably doesn't help you afford to be able to buy 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
; and "disallow" statements are to allow or
disallow various codecs. They way you've specified it above, you're
allowing a codec called xxx.xxx.xxx.0/255.255.255.0, which probably
isn't what you want.
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_
pers 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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e having Asterisk
play prompts or record calls 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.
we use. I really tried to figure 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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_
atus of the hint, and then when 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.
ed. (My gut feeling 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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hat. For a free download, check out
www.asteriskdocs.org.
There are obviously many other ways to do it.
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nstructions at
http://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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Jared Smith
Training
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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* R - channel group allocation round robin search backward
*
* c - Wait for DTMF digit to confirm answer
* r - Set distintive ring cadance number
* d - Force bearer capability for ISDN/SS7 call to digital.
*/
That's probably as definitive an answ
987654321
Message: Extension Status
Exten: 555
Context: lab
Hint: SIP/linksys
Status: 0
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f so, recent versions of Asterisk (1.6.0 and later, if I
recall) support SMDI.
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hips inside 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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id=15121 for more
details.
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act me directly for
more information.
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caller dialed extension 130. After the
first digit, the two patterns are tied. After 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 Opti
ownload a free PDF of the book at http://www.asteriskdocs.org/
or you can obviously buy a dead-tree version of the book from you
favorite bookseller.
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n
this list before, but I'd be happy to go over it again if anyone wants
me to.
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e tracker
(issues.asterisk.org) so that the developers can investigate further?
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in *many* situations, which is why I'm really
looking forward to doing more with it in the next few months.
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asterisk-users ma
'm assuming
that modern versions of Exchange still 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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that problem has been addressed in later versions 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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rompts, 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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nage
said prompts, but it does have the added benefit of supporting messages
in a variety of languages.)
Anyway, just my 2 cents (before taxes)...
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trailing spaces
You'll also want to remove any spaces from around the question mark
(after your expression).
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lopers spent a lot of time 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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-
the
problem actually 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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Bob's
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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Jared Smith
Tr
in chan_dahdi.conf on your live
system. Please note that the callprogress setting is *highly*
experimental, and in my experience causes more problems than it's worth.
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end up with 1.6.0.9 plus any bug fixes that have been applied since the
1.6.0.9 release.
Does that make sense?
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ill to me. 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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e switch type (I don't 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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ort, then Asterisk 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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Di
nd not necessarily being
officially endorsed by my employer.)
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on
before calling 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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I/g2/1646xxx... Did you mean to put those extra quotes in there?
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See doc/sip-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 Aster
you examined the output of "core show channels" to see what
application the hung channels are in? I'd start there.
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same server. Do 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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mas in your configuration. Try:
[incoming]
exten=> 1246463,1,Dial(SIP/8003,60,rT)
exten=> 1246463,n,Wait(5)
exten=> 1246463,n,Hangup
exten=> 6463,1,Dial(SIP/8003,60,rT)
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--
es but not
2bct... But for those few, I guess we can add yet another option.)
It seems silly to have to recompile just to get this functionality.
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d goes on its merry way. (If
a developer happens to read this and needs a pet project -- it would be
nice 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!
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T
re absolutely 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.
---
Jared Smit
use AGI to do this, you're pretty much limited 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.
l file is in a different location. Is Asterisk running? If
so, can you find the asterisk.ctl file that was created when it was
started?
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