t even attempting to send any video, then it likely
means that there is some other issue. It may be a bug, or it may be some
erroneous condition in the environment. Hard to tell yet though.
Mark Michelson
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for Asterisk to be able to open a single
file containing video and accompanying audio and be able to play those back.
Mark Michelson
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Check out the new Aster
On 07/10/2015 11:53 AM, Rodrigo Pimenta Carvalho wrote:
Hi.
The ASTERISK wiki has a page showing the function PJSIP_HEADER(). However, it
doesn't explain if such function works only over SIP INVITE messages or if it
can be use, for example, to read headers from others types of SIP messages
can
name them whatever you wish. The important piece of information when
determining what type of configuration section it is is the type=
option for the section. With no type= option set, the configuration
section is completely ignored.
Mark Michelson
and they said that this would be good to put
out on the -users list too.
Mark Michelson
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there are no ringing channels in context [from-my-sip-provider]
there are no calls to pick up there. However, since [context-100] and
[context-200] both have ringing channels, doing a call pickup in either
of these results in a successful pickup.
Mark Michelson
archive does not seem
to have my presentation video available.
Mark Michelson
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the CLI command agi show set
music
Mark Michelson
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voicemail on the file system, then you can just go to
/var/spool/asterisk/voicemail/context/mailbox/ and delete the items in
there
that you want to. The INBOX and Old folders contain new and old messages.
Anything else in there will be greetings and other similar recordings.
Mark Michelson
is used for both inbound and outbound calls.
Mark Michelson
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in SDPs and thus the port that
Asterisk will instruct the far end to send the media to.
Mark Michelson
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instead of creating the tag based off an already-fixed branch. This was
an oversight on our part, and we'll do our best not to make such a mistake
again.
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AstriCon
. Then,
based on the input, you can choose whether to run the Voicemail application.
Mark Michelson
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Actually, this feature is only in Asterisk trunk currently. It will be present
in 1.6.3 once it has been branched.
Mark Michelson
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Barry L. Kline wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Mark Michelson wrote:
You need to set a call-limit for the SIP peer. Device state calculation for
a
SIP peer is predicated on both the call-limit and busylevel. Let's say that
you
were to have a call-limit of 2
: not in use
1 call: in use
2 calls: busy
Basically, the busylevel defaults to the call-limit value. Now if you add a
busylevel = 1 to sip.conf, these are the device states reported:
0 calls: not in use
1 call: busy
2 calls: busy
Mark Michelson
DTMF. I would be willing to
bet that the other phones on your network are not using INFO for transmission
of
DTMF, and so they are not experiencing the same issue.
Mark Michelson
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possible
workarounds.
Thank you for your helpful and constructive criticism.
Mark Michelson
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application
Queue in the CLI.
What you need to look at is the queues.conf.sample file in the configs/
directory of the source. There you will find a myriad of options you may set
for
a queue, including the DTMF exit option you want.
Mark Michelson
to the 1.6 branches to improve the
spiral
support in Asterisk. If you are able to retry with the latest subversion
checkout of the branch you are using, you may find that things are working
better now.
Mark Michelson
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one
directory level and see if it plays, then.
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paused
since, once again, there is no monitor attached to the callee's channel.
So for the purposes of your setup, the only way you're going to be able to get
what you want working, short of actually changing the source code, is to only
allow the caller to be able to pause the monitor.
Mark
behave.
Mark Michelson
Thanks,
Josh Fuller josh.ful...@telus.com
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voicemail mailbox and entering an agent password).
Mark Michelson
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, the setting which forces media onto Asterisk during a SIP
call is the canreinvite setting.
Mark Michelson
Before I could call all my clients, I had musiconhold when putting 'on
hold' and I was just figuring out how parked calls worked...
Thanks for the help !
Jonas Kellens
to accept or reject
the
call.
The problem is that this does not allow for the calling party to also hear the
prompts.
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there was. Please upgrade to 1.4.24, where the problem has been fixed.
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problems, just a new
error message that points to problems that have been around a long time, most
of
which probably aren't that big a deal to begin with.
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thought that attempting to send zero-length data was
pointless and that if no data were passed to the application, it likely was due
to an error by the user.
Mark Michelson
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be found in doc/backtrace.txt in the Asterisk
source.
I suspect this is a regression introduced between 1.6.0.6 and 1.6.0.7 since
1.6.0.8 is exactly the same as 1.6.0.7, except for the security fix for
AST-2009-003.
Mark Michelson
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. Then type
queue show again and you'll see all the queues. I'm not sure why it was
written this way. If you use any 1.6 version of Asterisk, you will find that it
does not behave this way. queue show will always show all queues.
Mark Michelson
().
Why do you have the call-limit set to 1, anyway?
Mark Michelson
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Mr. James W. Laferriere wrote:
Hello Mark Miquel ,
On Thu, 26 Mar 2009, Mark Michelson wrote:
Miguel Molina wrote:
Hi all,
For those of you people that use Agents (with Agentlogin, not
AgentCallbackLogin) on a call center, I have this need: when the agent
logs in, a channel keeps
of the corresponding channel's callerid structure. Once
the
changes from http://reviewboard.digium.com/r/201 are merged into Asterisk
trunk,
then Asterisk will also generate a Diversion header if you have configured
Asterisk to generate redirecting information.
Mark Michelson
Olivier wrote:
2009/3/27 Mark Michelson mmichel...@digium.com
mailto:mmichel...@digium.com
Olivier wrote:
Hi,
Is anyone aware of SIP Diversion header ?
It seems currently supported by Comverse (formely NetCentrex)
softswitch
and some hardphones
show online which will only list
logged-in
agents.
Mark Michelson
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become unpaused after a certain time. If the member
is
ready to receive calls again before the time has expired, he can dial an
extension to unpause himself.
Mark Michelson
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Kevin P. Fleming wrote:
Mark Michelson wrote:
You can work around the bug, although it's not exactly optimal. What you can
do
is to modify your dialplan as follows:
exten = 301,n,Set(DYNAMIC_FEATURES=monkey)
Couldn't you just set _DYNAMIC_FEATURES here and have it get
automatically
to at least 4. That way I can hopefully
see what the problem is. Thanks.
Mark Michelson
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David Ruggles wrote:
I'm sorry, but it looks like it's working correctly now. I will update the
bug if I am able to verify any problems.
Thanks,
Heh, no reason to be sorry for it working :)
When you say it works now, was this with or without the patch applied?
Mark Michelson
David Ruggles wrote:
It was with the patch applied, but after I restarted asterisk.
Thanks,
Fix committed to Asterisk 1.4 in revision 181990.
Mark Michelson
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specify a full
channel name and either an app or an extension) and it also requires that one
of
the message or pdu fields of the outgoing call are filled in. You may want
to check bristuff documentation to figure out what these mean since they are
not
part of a regular Asterisk installation.
Mark
specifies the amount of time Asterisk should wait between DTMF
presses when you are dialing a feature code. So in your case, I'm guessing that
you pressed # but could not press 9 in time for Asterisk to recognize this
input
as part of the same feature.
Mark Michelson
if no timeout was given when calling the Queue application.
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sides can use the feature you have set.
This is a bug, and so there needs to be action to fix it correctly. What I've
suggested is just a workaround, but it should get you through your problem for
now.
Mark Michelson
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for me (TM). See if it
works out for you, too.
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the previous
version. In addition, you can check UPGRADE-1.6.txt to see about changes you
may
have to make when upgrading.
Mark Michelson
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to a version with DAHDI support. All variants of Asterisk 1.6 only support the
use of DAHDI.
Mark Michelson
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,Local/${EXTEN:3...@agentsSIP/${EXTEN:3})
Doing this will tell app_queue to use the SIP channel's device state to
determine if the member is available, but when it comes time to call the agent,
it will actually place the call to the local channel provided.
Mark Michelson
using? There was a recent bug introduced in
1.4.23. The fix for the issue is here:
http://svn.digium.com/svn-view/asterisk?view=revrev=174218
Mark Michelson
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Mark Michelson wrote:
Mike wrote:
Hi,
I have a customer running a 120 second long WAV file on their MoH. The
problem is that it's always starting from the beginning, so people being
put on hold, talked to, put on hold again, etc always hear the first
10-15 seconds
1.6.0 or higher, take a look at
configs/cdr_adaptive_odbc.conf.sample for some examples.
Mark Michelson
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= SIP/2000,Mark Michelson,3
In the above example, Mark Michelson is the name of a queue member who
can be reached by calling the interface SIP/2000. His penalty is 3. The
rule for penalties is that members with lower penalties are called
before members with higher penalties. If all the members
What version of Asterisk are you using? If you're using 1.4.23, there was a
confirmed problem which has been fixed now in the 1.4 svn branch. For the
issue,
please see http://bugs.digium.com/view.php?id=14206
Mark Michelson
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is broken too, SIP doesn't work on 2 difference boxes i tried it
on.
What's broken exactly? Saying SIP doesn't work is not a helpful description
of
what is going wrong. Are there any open bug reports that describe the problem
you are having?
Mark Michelson
and see
what I can find out.
Mark Michelson
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is in use then ringinuse should work correctly. If the
device state is reported as anything other than in use then that would
indicate that IAX2's device state reporting may be inaccurate and also
ringinuse
will not work.
Mark Michelson
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to ringing instead
of
confirmed if a phone is ringing. Also, Asterisk will place a direction
attribute inside the dialog XML tag in this situation too.
The short version of this is that the notifyringing option will specify a
ringing state in NOTIFY messages but only for certain types of phones.
Mark
option and it will dump a core file if it should crash.
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the queue member with information about the incoming
call.
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?
Thanks!
A quick look at the code and your config leads me to believe you're doing
everything correctly. What version of Asterisk are you using? Are you using
realtime queues/queue members?
Mark Michelson
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Mark Michelson wrote:
Christopher Aloi wrote:
Hey List,
Anyone know the correct way to override an announcement on a queue by
queue basis?
My goal is to have one of my queues say press one to blah.. and no
position announcements I have the jump from queue context working (the
press 1
in the
sounds directory.
Mark Michelson
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Christopher Aloi wrote:
Yah - Found my problem, I can't spell -
periodic-*annouce* = SD-PLS-HOLD
periodic-announce-frequency=10
: )
Oh, Ha! That'll do it every time.
Mark Michelson
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. That is the response given to a Command action
assuming that a command was provided and the command is not blacklisted.
Mark Michelson
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reinvites to the endpoints even if you
have configured chan_sip to allow reinvites to be sent. Other factors which can
contribute are use of applications like Monitor and MixMonitor which require
the
media to go through Asterisk.
Mark Michelson
getting this documented better.
Mark Michelson
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,
MixMonitor recordings can be set to survive a transfer if you are using
Asterisk
1.4.23 and make use of the AUDIOHOOK_INHERIT function. For more information on
its use, you can issue the command core show function AUDIOHOOK_INHERIT from
the Asterisk CLI.
Mark Michelson
should report it
as
a bug on bugs.digium.com. Be sure to attach a backtrace from the crash as
described in doc/backtrace.txt in the Asterisk source.
Thanks,
Mark Michelson
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or error condition. This functionality was
marked deprecated in Asterisk 1.2. An option to disable it was provided with
the default value set to 'on'. The default value for the global priority
jumping option is now 'off'.
Mark Michelson
do not have a
copy of the book, the following web page is listed as containing errata,
examples, and any additional information:
http://www.oreilly.com/catalog/9780596510480
Mark Michelson
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.
We can't fix what's wrong if we don't know what's wrong to begin with. :)
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are
t38UDPFEC and t38UDPRedundancy, with the former being the default. Looking at
the code, it appears that the options are case-sensitive for udptl.conf, which
is quite a bit different from the rest of Asterisk, so be sure to get the case
correct.
Mark Michelson
those
return values and how recently that was fixed, it's probably something you can
ignore. Of course updating to a more recent checkout of Asterisk will clear
such
warnings up.
Mark Michelson
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, but as far as I know, that should be corrected in later versions. What
version of Asterisk are you seeing this with?
Mark Michelson
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. There is no
minimum or maximum limit to what this string may be.
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minority or if most
subscribers to the -announce list would appreciate seeing such messages.
Mark Michelson
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be that there is something obviously malformed in
the
SIP requests being sent by Asterisk.
Mark Michelson
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Kevin P. Fleming wrote:
Mark Michelson wrote:
If you are using gsm prompts and gcc version 4.2 or higher, then you may be
experiencing the optimizer bug that gcc has with gsm audio. The workarounds
for
this are to use a different format for sounds or to set the DONT_OPTIMIZE
flag
probably what you searched for. There
have
probably been hundreds of threads on that subject on this list, so filtering
through it all is not easy.
Mark Michelson
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. If you want an optimized build and gsm formatted sounds, then
you
could always attempt downgrading your gcc version to 4.1 or earlier.
Mark Michelson
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through it.
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/view.php?id=14151
http://bugs.digium.com/view.php?id=14153 needs to be reopened.
Philipp Kempgen
I re-opened this bug. Thanks for bringing this up.
Mark Michelson
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to worry about is that warning. If you find
the warning to be annoying, the best I can offer you is to either not log
warnings (a bad idea, imho) or just remove that line of code from the source.
Mark Michelson
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get more details on this by
issuing the command core show application Chanspy in the Asterisk CLI.
Specifically look at the g option.
Hope this helps.
Mark Michelson
Mateusz Pawlowski wrote:
Hi,
I was asked to create a Queue which instead of playing MoH it generates
the ringing tone. I had a look around but could find anything, I would
welcome and help.
Regards
Mateusz
You can pass the 'r' option to the Queue application for this purpose. As an
they are back
up.
Happy Holidays,
Mark Michelson
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conditions in the
queues.conf.sample file in trunk's configs/ directory in the source. Since this
feature is already in trunk but not in any released version of Asterisk, it
will
be present in Asterisk version 1.6.2.
Mark Michelson
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for your patience.
Mark Michelson
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it.
The wiki is no help on that one…
Mike
If you look at the help text for the Directory application using the Asterisk
CLI (core show application directory), it specifies that pressing the '*' key
will send you to the 'a' extension if it exists.
Mark Michelson
?
Thanks in advance
--
Pagarbiai / Best Regards,
Giedrius Augys
This is a bug you are experiencing, which I fixed recently in a series of
commits. Assuming you are using a 1.6 tag, the next build should have this
problem fixed.
Mark Michelson
lately due to more pressing matters.
Mark Michelson
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packets in a
program like wireshark and analyze them there as well.
Mark Michelson
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to
reloading and resetting queues. The branch is located at the following URL if
you wish to give it a test:
http://svn.digium.com/svn/asterisk/team/mmichelson/queue-reset
If you run the code there, you'll find that there is a command called queue
reset stats which should do what you want.
Mark
hello and a variable
called BAR being set to world.
Mark Michelson
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Mark Michelson wrote:
Philipp Kempgen wrote:
How is MSet() different from Set()?
Is it supposed to be a Multi-Set()?
Why was it added in 1.6?
Philipp Kempgen
It is a Multiset application. My recollection of the addition is that due to
parser changes in 1.6, a statement like
that
anonymous_call_rejection and 22 were supposed to be arguments to the gosub, and
not treated as a label. Since no extension exists with that label, that is why
the gosub is now failing.
This is definitely a bug and needs to be corrected before the next version of
1.6.0 is released.
Mark
: callandqueue
Message: Channel not specified
Is anyone else seeing anything like this?
Thanks for pointing this out. I have located the erroneous code and have fixed
it in subversion, revision 161490. The next rc of 1.6.0 will not have this bug.
Mark Michelson
Mark Michelson wrote:
Gary Hawkins wrote:
Hi all,
I've just upgraded to latest 1.6.0 SVN from a few days ago and my Gosubs
have stopped working.
This is from the verbose logs:
-- Executing [EMAIL PROTECTED]:4] GotoIf(IAX2/aaisp-3802,
1?5:7) in new stack
-- Goto (incoming-aaisp
have no Digium hardware, but I still need the ztdummy timer (or
whatever it`s called now). How do I get myself going?
Regards,**
* *
*Mike*
DAHDI has 'dahdi_dummy' in place of ztdummy. You should be able to use it
exactly the same way that you used ztdummy.
Mark Michelson
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