Tilghman Lesher wrote:
On Wednesday 03 October 2007 06:09:01 Peter Fern wrote:
Of course, I could be missing something obvious, please correct me if
that's the case.
I invite you to try it. You could make a lot of really smart people look like
fools if you're able to mix
Tilghman Lesher wrote:
Or, in other words, you cannot mix compressed data. You must first
decompress the data for mixing, then recompress it for transmission.
During both operations, there is a potential for signal degradation.
Ummm, why?? Unless you can explain some technical reason
searched the bugtracker to no avail, full debug gives no useful
data that I can see - is this a known bug, and does anyone have a
workaround?
Regards,
Peter Fern
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing
Scott, you know you really have too much time on your hands when... :)
Scott Gifford wrote:
This extremely useful dialplan requires the standard Asterisk sounds,
plus the additional ones in the asterisk-sounds package.
Scott.
[haiku]
exten = s,1,Playback(privacy-please-dial)
exten =
${EPOCH} * Current unix style epoch
Add your 5mins as seconds, and convert if necessary, you could do it
like this in the dialplan to give the same format as ${DATETIME} (which
is deprecated by the way):
${STRFTIME($[${EPOCH} + 300],,%d%m%Y-%H:%M:%S)}
Read doc/README.variables to find out
Yeah, I got it a couple of times.
Doug Lytle wrote:
An outside caller started to leave voice mail.
The CLI shows:
Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/sip/4232/INBOX/msg format: gsm,
0x8295d40
-- x=1, open writing:
You should use the 'filename' parameter of Read to play the audio so
that it captures the input. Currently what's happening is that digits
entered whilst background is running are passed into the dialplan
context, since there's no match in the dialplan and you don't have an
'i' extension it
Probably because the Local proxy channel drops out once the two sides
have been bridged. If you want the Local chan to stay up, use the /n
parameter and the local channel won't perform the native transfer. This
does have it's own problems, but should do what you want.
eg:
Channel:
From:
http://www.voip-info.org/wiki/view/Asterisk+cmd+AgentCallbackLogin
More info
Unlike with AgentLogin the agent is not permanently off-hook (on-line).
Instead the agent will be called at the designated extension when a new
queue caller has been assigned to him. The agent goes off-hook
own
variable (shortterm).
On 4/20/06, Peter Fern [EMAIL PROTECTED] wrote:
Probably because the Local proxy channel drops out once the two sides
have been bridged. If you want the Local chan to stay up, use the /n
parameter and the local channel won't perform the native transfer. This
does have
Steve Kennedy wrote:
Is there a way to differentiate between a SIP address which hasn't
registered (but is within sip.conf) and one that's not there at all
(i.e. not in sip.conf) using a straight dialplan.
I'd like to differentiate actions depending the state of a SIP device
and whether it's
Douglas Garstang wrote:
The 'sip show channels' and 'show channels' command aren't exactly easy to
interpret, especially if one of the numbers has pic codes and rate centers
inserted (the rest is truncated on the output), or you have a proxy involved in
the call. Wish someone with some C
Tomislav Vojvodic wrote:
If you put 777 on some file, that means that anyone can read/write/execute
that file.. I think that file ownership isn't important in that case, but if
you want to set 'precise' permissions so that only user 'asterisk' can deal
with it.. then type
chmod 644 filename
Chris Mason (Lists) wrote:
I have been experiencing dropped calls on my iax2 connections between
my Asterisk server and my ITSP providers, I use Teliax and Voxee but
it seems to happen on both so I don't think it is the provider. I
don't see any packet loss at the time so I don't think it
Should be posted to the -biz list?
Steve Ducat wrote:
CHINA DID
I am once again in search of China DID's. Either Shanghai (021) or
Guangzhou (020).
Please advise if you can supply.
Steven Ducat.
___
--Bandwidth and Colocation provided by
I've had terrible audio quality with IAX2 on Asterisk 1.2, there was a
thread about it some time ago and many people reported the same thing.
I think the thread was entitled 'problem with new jitterbuffer
implementation' or something similar. However, it did not appear to be
the jitterbuffer
I've had the same problem with all boxen running the same version. We
ditched IAX2 for SIP and it has been working fine since.
Doug Lytle wrote:
Barry Flanagan wrote:
Hi,
I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to
connect to a 1.2.5 box for PSTN. There are 15
asterisk 1.0.9 ish. So perhaps
the IAX issue is only recent in 1.1-- ?
On 3/20/06, Peter Fern [EMAIL PROTECTED] wrote:
I've had terrible audio quality with IAX2 on Asterisk 1.2, there was a
thread about it some time ago and many people reported the same thing.
I think the thread was entitled
Still, if you mirror them yourself, this problem all but goes away.
Alistair Cunningham wrote:
Colin,
Because having the logic is not the correct thing to do from an
engineering point of view. Consider:
- What if Digium change the directory structure again? Having a
published directory
Use of '_.' is discouraged. In this case, '_[*X].' should work I think
Douglas Garstang wrote:
I'm trying to find a way in extensions.conf to match ANYTHING dialled,
including characters such as *.
The following works for numbers...
exten = _X.,1,AGI(script)
but doesn't catch when
I had exactly the same experience running IAX2, but also experienced
half-duplex calls on top of that (though I think that's a different but
with IAX handoff), and in the end dropped it completely for SIP.
We run g729 over dedicated fibre, and the resyncs were occurring all
over the place
In queues.conf:
[queuename]
announce-frequency = XX ; where XX = number of seconds
Rajkumar S wrote:
David Ankers wrote:
Don't you need an
exten = s,1,Answer
The full sequence is:
[ivr] ; Voice Menu
exten = s, 1, wait(2)
exten = s, 2, Answer
exten = s, 3,Goto,MainMenu|s|1
[MainMenu]
We disabled call waiting to stop the beep.
Gary Richardson wrote:
Hey,
I'm testing out some call queues. I have 7940's and 7960's with the
SIP 7.4 image.
I have a queue that looks something like:
[testqueue]
strategy = rrmemory
timeout = 15
retry = 5
weight = 0
announce-frequency = 0
Use a context=blah line in your queue config with an extension * in it?
Joseph Rothstein wrote:
I am trying to exactly this using 1.2.4, and it doesn't happne.
DTMF works fine for VM and IVR.
Joe
___
--Bandwidth and Colocation provided by
You can enable this on a per-peer basis with:
sip peers:
canreinvite=yes
iax peers:
notransfer=no
Check the iax.conf.sample and sip.conf.sample files for usage.
Nitin Gupta wrote:
Hi I was wondering if its possible to make Dial command bridge two
channels and after bridging bypass asterisk,
It errors when you ask it for the channel 'test-1' because the parameter
is the channel name, not the peer name. I've used 'show channels
concise' and then parsed the output in the past.
Peter Hoppe wrote:
Hello!
I have an asterisk setup where several sip devices are connected to an
The 'local' technology might be what you want, you can use it to dial an
extension/context, and handle call direction there. It will actually
always answer technically though, and will split your CDRs. You will
also want to watch out for variable inheritance when using local channels.
Brian
You could give each of your users a 'home' asterisk machine by an
extension numbering convention and route voicemail based on their
number. It would mean that voicemail messages won't be able to be
forwarded to a user homed on another machine though I would imagine. We
went for a centralised
Just set one in the dialplan as you enter the queue?
Damon Estep wrote:
I am using agentcallbacklogin for queues
I have a desire to modify the call behavior to the agents extension if
the call is from a queue (opposed to from a PRI or another extension).
Is there a channel variable that can
Only using SCCP, SIP firmware is set in stone.
Alex Ongena wrote:
Hi,
We are using Asterisk 1.2.1 with Cisco 7940 and 7960 phones.
Most things are running fine ;-)
But, when you are calling and you want to Transfer, you need
to press first on the 'more' button (4th), then you have the
label
http://www.voip-info.org/wiki/view/Iperf
Not just latency, but jitter, etc - basically you can simulate various
types of traffic and generate statistics. You will need two ends to
test with.
Cosmin Prund wrote:
As the subject line says: Is PING a good indicator of network latency? If
not,
in your kernel.
Mark Phillips wrote:
How does the customer maintain the message if I have to capture it
every time he changes it?
This is not the solution.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Peter Fern wrote:
Using the classic MoH, use a custom moh player (see
http
Don't know why you would have been experiencing pauses beforehand, but
you can specify a digittimeout with:
Set(TIMEOUT(digit)=#)
Michaël Gaudette wrote:
Hi,
I have a little situation with my dialplan, and I am wondering if what
I want is even possible.
Here it is: I have three
I'm pretty sure I've seen some commits dealing with channel locking
since 1.2.1
Brent Torrenga wrote:
Might it be related to the memory leak bug? Upgrade to 1.2.4? (shot in the
dark, a brainstorm on my part is all)
Here's what the logfile shows. Any ideas? And is
there a way to fix the
_X. should do the job for you.
Harald Holzer wrote:
Hi,
is there a way to executing commands in the dialplan regardless which number is
dialed before
the pattern matching starts ?
when a call enters the first context it would be nice if i can set some
variable or manipulate
a callerid, or
Using the classic MoH, use a custom moh player (see
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf)
and sox with the alsa pseudo-filetype, and output to stdout with the
correct bitrate and samples... see the sox manpage for instructions.
Untested, but I think
This all depends on what your existing setup is, mainly where you are
storing your sip users in the first place. No point duplicating your
user list. The Ciscos consume XML, so just parse out the list of users
via some scripting language from the DB/directory/flat-files or whatever
you're
Yep, I just got stung by this too - an hour of extreme pain, multiple *
boxen all failed at precisely the same moment, and they're in different
timezones, so must be a calc on epoch or UTC.
Anyone shed any light on this? I'm hacking our CDRs currently to work
around the difference in year,
That's been happenning since 1.2 - if the parent * process dies the
mp123 process doesn't go with it like it should and will just eat your
processor right up, if you're using 1.2, I'd definitely suggest using
native moh.
Robert La Ferla wrote:
Chris Albertson wrote:
Second even if there
Either include the context containing the definition:
[second-context]
include = other-context
or specify the context in the dial command:
Dial(local/[EMAIL PROTECTED])
Matt wrote:
Hi,
What do I have to do to get local\number to work in a context?
It works from my [from-internal]...
http://www.voip-info.org/wiki-Asterisk+FastAGI
Mike Fedyk wrote:
Is there anything like FastCGI for Asterisk so that AGIs can have
persistent processes?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
Just implemented a similar feature here - apparently the chanprefix
won't accept a full channel identifier, so I ended up dropping the last
character (this works for me since all the sip delivery we want to
monitor is to individual handsets - I won't be monitoring any channels
that are
Ahh, I'm running r7233, I'll update to the latest rev to pull in the
changes, thanks Juan.
Juan Jose Comellas wrote:
This problem has been already corrected in Asterisk 1.2. See this bug:
http://bugs.digium.com/view.php?id=6009
On Monday 09 January 2006 00:51, Peter Fern wrote:
Just
43 matches
Mail list logo
