James FitzGibbon wrote:
On 7/26/07, *James FitzGibbon* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Is it possible for qe.parent-membercount to be set to zero in a
queue where all agents but one are on the phone and that one
remaining agent lets their phone ring without
to the reporter's system, but I want to be
sure that this isn't standard behavior on certain hardware, OS, and
compiler combinations.
Are there any other users who have had this issue or one similar, and if
so what OS, hardware, and compiler are you using?
Thanks,
Mark Michelson
Tim Groeneveld wrote:
I am running r79979 of Asterisk Trunk, and I am having problems trying to use
app_queue.so.
I want to use the extension 510 to be a line where users can call technical
support.
Extensions 511 and 512 are used by the operators to dynamically make
themselves a Queue
Tim Groeneveld wrote:
On Tuesday 21 August 2007 12:32:12 am Mark Michelson wrote:
When users call 510 then, it actually does ring everyone who has called
511.
The problem is when the operator comes to pick up the call. The operator
hears nothing, and the user still hears the Music
, you stay in the queue.
Have you added additional queue members besides the ones you specified
in queues.conf?
Mark Michelson
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Vieri wrote:
Hi,
I setup a queue (number 4050) with one static agent
(extension 4054).
What I would like is that when someone calls the 4050
queue and there are neither dynamic agents logged in
nor is the static agent 4054 on-line then the caller
gets out of the queue and falls into
Mark Michelson wrote:
Sander Smeenk wrote:
Quoting Mark Michelson ([EMAIL PROTECTED]):
| app_queue.c: No one is answering queue '511' (7/2/0)
Have you added additional queue members besides the ones you specified
in queues.conf?
Yes. There's
Sander Smeenk wrote:
Quoting Mark Michelson ([EMAIL PROTECTED]):
| app_queue.c: No one is answering queue '511' (7/2/0)
Have you added additional queue members besides the ones you specified
in queues.conf?
Yes. There's a number of dynamic members that logged
,warning,error,debug
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through the process of changing
their PIN, recording their name, and their greetings.
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as opposed
to a separate option.
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to a high value so it
is not put into action) and also make sure that both inbound
and outbound calls are accounted for.
Example:
[general]
limitonpeer = yes
[peername]
type=friend
call-limit=10
Mark Michelson
81120. If you are
running a version of Asterisk prior to this revision and using ackcall,
I'd suggest upgrading and seeing if the issue still exists.
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of a stretch to use it based on what's provided in the
book.
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is
over, I'd suggest moving the System command to the h extension. The h
extension is called on hangup, so it should clear up your issue.
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not sure how the gain option works as an argument to Voicemail(), but I
know
that the volgain option for e-mail attachments requires that you have sox
installed in order to work properly. If you don't already have it installed, I
would suggest installing sox and seeing if that helps.
Mark
penalty, and a boolean column for
determining if the member is paused.
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http://www.kolmisoft.com
For those following this issue, there was a bug filed for this (issue #12362:
http://bugs.digium.com/view.php?id=12362) and it has been fixed, too (Asterisk
1.4 svn revision 113240).
Mark Michelson
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and not just set the call-limit in the
general section.
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of active channels would
always end up returning the first channel it found. If that happened to be a
spy-able channel, then great, otherwise you'd never spy on anything.
Mark Michelson
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The Asterisk development team has released Asterisk-Addons version 1.2.7,
1.4.9,
and 1.6.0-beta4 to address a major security vulnerability in the ooh323 channel
driver. The releases may be downloaded from http://downloads.digium.com/.
AST-2008-009 details a remote crash vulnerability in the
with the JOINEMPTY
queue status.
With regards to why the member is showing up as invalid, I would assume it is
because the member's interface is set to 9001 instead of something like
SIP/9001.
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by asterisk when their
queue is called.
Any ideas?
not in use is just the current device state of that queue member. Members who
are not in use should be called by the queue. Do you see anything indicating
an error on the console when you try calling?
Mark Michelson
.,1,Set(CHANNEL(language)=pl)
exten = _078.,2,Wait(2)
exten = _078.,3,Playback(platna)
exten = _078.,4,SetAMAFlags(documentation)
exten = _078.,5,Dial(SIP/voipnet/0048${EXTEN:1},120,Tt)
exten = _078.,6,Hangup
exten = _078.,3,Playback(platna|noanswer)
Mark Michelson
to 1.6.0.
Asterisk 1.6.0 supports realtime music on hold.
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to hangs up.
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gave would make sense if there
are no members in the queue, no callers had been handled yet, and there was no
weight or maxlen set in queues.conf. I'm not sure if that's the case in this
scenario or not.
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the 't' or 'T' flags to
the options for Dial().
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phone number calling in.
Thanks!
jlc
The To: tag on the response is the tag generated by your phone. It is generated
pretty much at random. It's just a happy coincidence that it happened to nearly
spell the word faulty. Still, that's kind of funny though :)
Nothing to worry about.
Mark
codec settings.
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jiangtao wrote:
I'm using asterisk 1.4.21 and a problem with sip reg server
In SIP.CONF
register = 07070480800:[EMAIL PROTECTED]
register = 07070480801:[EMAIL PROTECTED]
register = 07070480802:[EMAIL PROTECTED]
register = 07070480803:[EMAIL PROTECTED]
register = test1:[EMAIL
the DYNAMIC_FEATURES variable? Something like:
exten = blah,n,Set(DYNAMIC_FEATURES=automon)
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is: http://bugs.digium.com/view.php?id=12924
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the message is going to agent1 and agent2 who
actually takes the call never sees the message
What type of channels do you use for your agents? If you're using Agent
channels
(the type which are configured in agents.conf), are you logging them in using
AgentCallbackLogin?
Mark Michelson
for the queue member has become stuck. What types of channels do you
use
for your queue members?
Mark Michelson
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at all.
In order to figure out why there is one-way audio, you would need to provide a
sip debug of the call. Based on the fact that you have nat=yes for both SIP
friends, I'm guessing that there's some sort of NAT issue here, but I can't be
certain.
Mark Michelson
ways for a
member to become automatically paused.
That being said, it could be that you have discovered some sort of bug in 1.4.
When does this appear to happen? Does it happen randomly or is the situation
reproduceable?
Mark Michelson
the sip.conf option.
Mark Michelson
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into congestion and I get a busy tone. What could I be
doing wrong?
James
As a sanity check, you may want to place a NoOp(${EXTEN}) prior to the dial. If
you set the verbosity high on the Asterisk console, then you can see what the
value of EXTEN is when the NoOp occurs.
Mark Michelson
to be Unavailable
until
qualify determines that the phone is available.
Ah..I forgot to say I do not use agents but only static queues, no real
time stuff.
Giorgio
Mark Michelson
Mark Michelson wrote:
Giorgio Incantalupo wrote:
Hi Mark,
it is show queues I use to see if phones
in 1.4.21.2 are the two IAX2 security vulnerability
fixes mentioned in AST-2008-010 and AST-2008-011.
I believe the bug that you want is here:
http://bugs.digium.com/view.php?id=12921
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SIP wrote:
Is there a way to set a call timer on calls created with call files? I'm
looking specifically at having Asterisk hang up the call after a certain
period of connection.
Obviously, when I try passing an |S(time) on the channel line, I get an
invalid call file... so I'm
Mark Michelson wrote:
SIP wrote:
Is there a way to set a call timer on calls created with call files? I'm
looking specifically at having Asterisk hang up the call after a certain
period of connection.
Obviously, when I try passing an |S(time) on the channel line, I get an
invalid call
yet which has the fix, but it
will be in the upcoming 1.4.22 release. If you are using a subversion checkout
of version 1.4, then you can update to any revision after 132790. Hopefully
you'll see that the spirals work correctly.
Mark Michelson
, then a
different fix may be in order instead.
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options and
then recompiling Asterisk. I also believe that you can set the optimization
level for compilation to -O2 in Makefile.rules and have no choppy audio, but I
cannot confirm this.
Of course, if this server isn't running GCC 4.2, then you can ignore everything
I've said so far :)
Mark
of SIP, this would indeed
mean that there is a 500 ms delay between receiving the 200 OK from the callee
and sending a 200 OK to the caller.
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scripts provided for Asterisk, including safe_asterisk, will dump core to /tmp.
Also, Fedora Core 2?! Ouch! ;)
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Regards,
Giedrius Augys
You can check the value of QUEUESTATUS after the call to Queue(). If the call
timed out, then it will be set to TIMEOUT.
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much better now, and the AutoFill parallel
call distribution is really nice.
Bob
I'm glad to hear that you're enjoying Asterisk 1.4! Unfortunately, the
shared_lastcall option is only in versions 1.6.0 and up.
Mark Michelson
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and you'll be stuck with the behavior you're seeing.
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or is not being generated in the first place. Seeing a SIP debug from the call
would help to diagnose the problem.
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for the downloadable pdf:
http://downloads.oreilly.com/books/9780596510480.pdf
Here's a link for the book in html format
http://tfot.leifmadsen.com
Best of luck to you!
Mark Michelson
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Andres wrote:
Hi,
I was trying to get more details on some deadlocks we are experiencing
with Asterisk Version 1.4.21.1 so I went ahead and compiled it with
DONT_OPTIMIZE and DEBUG_THREADS.
The problem now is that asterisk_addon modules will not load. I get
messages like:
[Sep 16
number as an
argument to the application if you don't want to be prompted for a mailbox
number.
Mark Michelson
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the
INVITE that Asterisk is sending.
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legal.
The real question is why does Asterisk think it is the same request
when the
from tag is different ?
b.
Asterisk ignores tags in To and From headers unless you have pedantic=yes set
in sip.conf.
Mark Michelson
missing something?
Thanks in advance
David
The AMI atxfer command is not in 1.6.0 and will be in no releases of 1.6.0. It
is in 1.6.1, for which a beta is currently available.
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than the default. checkmwi defaults to
checking the server for new messages every 10 seconds. If you set this to be
something like 60, this should ease the load somewhat.
Mark Michelson
, you can issue the command sip
set debug in the CLI. Then all SIP messages will be written anywhere where you
are logging verbose messages.
Mark Michelson
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noticed earlier this morning that the SMTP daemon was not
functioning properly at carolina.digium.com, and so there was a big backlog of
queued messages. Part of correcting the problem is that all these queued
messages are finally being sent out.
Mark Michelson
to be located in North America to use
an Open Source PBX that is located in North America. Of course, since I didn't
come up with this survey, I have no idea what the actual intent of the question
was.
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Yann JOUANIN wrote:
Hi all,
I have a strange problem when using IMAP storage.
I have the error : Couldn't find mailbox default in context default just
after the caller listen the unavailable announce.
The communication is broken then.
I added in the voicemail.conf the mailbox default,
[EMAIL PROTECTED] wrote:
Hi,
I'm playing around with the QUEUE_WAITING_COUNT function but it always
seems to return zero? I've tried everything. I suspect that this feature
is not implemented in 1.2.7 which I am running..
Does anyone know in which version this function was added?
.
If things still fail after those two steps, then respond with the section from
the config.log file which displays the failure that occurred when searching for
imap support.
Mark Michelson
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c james wrote:
Mark Michelson wrote:
c james wrote:
I have c-client installed on a 64bit system running Gentoo. I am trying
to run configure so I can test the IMAP voicemail functionality. But
asterisk-1.4.22 # ./configure --with-imap=/usr/include/imap
just gives me the following error
c james wrote:
Mark Michelson wrote:
c james wrote:
Mark Michelson wrote:
c james wrote:
I have c-client installed on a 64bit system running Gentoo. I am trying
to run configure so I can test the IMAP voicemail functionality. But
asterisk-1.4.22 # ./configure --with-imap=/usr/include/imap
c james wrote:
Mark Michelson wrote:
c james wrote:
Mark Michelson wrote:
c james wrote:
Mark Michelson wrote:
c james wrote:
I have c-client installed on a 64bit system running Gentoo. I am trying
to run configure so I can test the IMAP voicemail functionality. But
asterisk-1.4.22
channels first, so if your outgoing calls start with the higher
numbers
it decreases the chance for glare on calls.
Mark Michelson
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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
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
hello and a variable
called BAR being set to world.
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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
packets in a
program like wireshark and analyze them there as well.
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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
lately due to more pressing matters.
Mark Michelson
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?
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
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
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.
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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
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
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.
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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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through it.
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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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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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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
be that there is something obviously malformed in
the
SIP requests being sent by Asterisk.
Mark Michelson
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minority or if most
subscribers to the -announce list would appreciate seeing such messages.
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. There is no
minimum or maximum limit to what this string may be.
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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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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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