channels first, so if your outgoing calls start with the higher
numbers
it decreases the chance for glare on calls.
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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
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
.
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.
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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
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.
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
, 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.
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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
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.
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
number as an
argument to the application if you don't want to be prompted for a mailbox
number.
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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
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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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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and you'll be stuck with the behavior you're seeing.
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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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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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scripts provided for Asterisk, including safe_asterisk, will dump core to /tmp.
Also, Fedora Core 2?! Ouch! ;)
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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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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
, then a
different fix may be in order instead.
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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
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
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
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
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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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)
Mark Michelson
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to
is: http://bugs.digium.com/view.php?id=12924
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codec settings.
Mark Michelson
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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
the 't' or 'T' flags to
the options for Dial().
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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.
Mark Michelson
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to 1.6.0.
Asterisk 1.6.0 supports realtime music on hold.
Mark Michelson
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.,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
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
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.
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
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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and not just set the call-limit in the
general section.
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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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penalty, and a boolean column for
determining if the member is paused.
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remotely, then you can use the
queue add member command (add queue member if you're using 1.2) from there
as well.
2) How do you keep those phones in that queue even after the system reboots?
set persistentmembers=yes in queues.conf
Rob
Mark Michelson
Atis Lezdins wrote:
On 1/17/08, Mark Michelson [EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
Hi,
I'm wondering - why SIP device state doesn't get updated to anything
else, except Not In Use.
For queue call (with Local channel) i get:
app_queue.c: Device 'SIP/21168' changed to state '1
Mark Michelson wrote:
Atis Lezdins wrote:
On 1/17/08, Mark Michelson [EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
Hi,
I'm wondering - why SIP device state doesn't get updated to anything
else, except Not In Use.
For queue call (with Local channel) i get:
app_queue.c: Device 'SIP/21168
call-limit in the general section. This setting, however, may
only be set per peer (or user). Unfortunately, there's no warning message
output
if an unrecognized option is set in the general section.
Mark Michelson
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Andrew Joakimsen wrote:
Anyone else have issues with T.38 where the call drops after T.38 is
attempted to be negotiated, with a message like the below?
WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in
c= line, 'IN IP4 100101'
The problem is that 100101 is neither a
Stefan Guenther wrote:
Hi,
I have a question about the definition of agents.
The agents.conf file looks like this:
[general]
persistentagents=yes
[agents]
maxlogintries=5
ackcall=no
wrapuptime=500
musiconhold = default
group = 1
agent = 1311,1311,Tom
agent = 1531,1531,Tim
, Asterisk 1.4.15, FreePBX 2.3.1
The option you are looking for is called serveremail. By default, if this is
not set, it will be set to asterisk. Set this in the [general] section of
voicemail.conf.
Mark Michelson
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to a change in the way autoservice is handled. I think
that
if you were to upgrade to SVN revision 90432 or later, this error will not
occur.
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Anthony Messina wrote:
On Thursday 06 December 2007 12:42:42 pm Mark Michelson wrote:
Anthony Messina wrote:
After updating to 1.4.15, I have the following issue:
When I try to use the M macro option in the Dial() option, I get the
following in the console:
-- Executing Dial(Zap/1-1,
Zap
(conn^1002));
and it should work. Notice that the pipe between the L and M options has been
removed.
Mark Michelson
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[EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
Short of replacing a sound file with a sound file containing only a
short period of silence, is there any way to suppress certain sounds
from playing during queue processing by configuring for example
queues.conf or other similar files?
Which
[EMAIL PROTECTED] wrote:
Short of replacing a sound file with a sound file containing only a
short period of silence, is there any way to suppress certain sounds
from playing during queue processing by configuring for example
queues.conf or other similar files?
Which announcements are you
, not fx0_ks.
Mark Michelson
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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
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.
Mark Michelson
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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.
Mark Michelson
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asterisk
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
as opposed
to a separate option.
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through the process of changing
their PIN, recording their name, and their greetings.
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,warning,error,debug
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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
, you stay in the queue.
Have you added additional queue members besides the ones you specified
in queues.conf?
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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
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
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
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
[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?
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,
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