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On 12/11/2011 10:59 PM, Mike Diehl wrote:
Should I go to 1.8.x? Or all the way up to 10.x? This is a
production system and I can't afford to be testing code.
The 1.8 series is the current LTS release.
Barry
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On 08/05/2011 04:32 AM, Richard Zulu wrote:
I would like to import my dialplan into freepbx+asterisk since I am
switching to that...how can I create my own custom dialplan in
freepbx?
I'm not sure why you'd want to... freepbx is anathema to
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In
http://svnview.digium.com/svn/asterisk/branches/1.8/CHANGES?view=markup,
line 180 states:
Voicemail now runs the externnotify script when pollmailboxes is
activated and notices a change.
My voicemail.conf configuration for my LDAP vm storage is
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On 07/28/2011 02:42 PM, Barry L. Kline wrote:
Is there some other parameter required to get this to fire or am I
reading more into that sentence from the CHANGES document than is
actually there?
Sorry for replying to my own post, but I've done
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I found what I believe to be a bug and have submitted it:
https://issues.asterisk.org/jira/browse/ASTERISK-18207
Please correct me if I'm wrong.
Barry
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On 06/21/2011 08:37 AM, Jonas Kellens wrote:
At the moment, I don't really know what I'm looking for. So if anyone
knows how to do it in a Cisco, Grandstream, Yealink or Snom IP-phone I
can find out myself what settings to look for in other
On 02/01/2011 12:34 PM, Harel Cohen wrote:
As one with theoretical knowledge in programing, but never on Linux, I
can understand terms and code structure but I don’t know:
1. What shell commands (e.g. ./configure, make, make install etc.)
should I run to recompile Asterisk (same version)?
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Bryan Jacobs wrote:
I wonder if all the cell providers let you do this?
I presume you mean turn off voice mail. I don't know, but the first
time I called Verizon to have it done the gal I spoke with said it
couldn't be done. So I said thanks,
Bryan Jacobs wrote:
I can't just call the car - the car is my cell phone DID with a
bluetooth kit.
I did this same thing you're attempting. I have a desk set at home, a
Polycom in my office and my cell phone all being called at the same
time. I called Verizon and had them disable voice mail
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DHAVAL INDRODIYA wrote:
where as I am using Asterisk 1.6.0.5 and my machine is using
*safe_asterisk* script asterisk running
Why are you using such an old version in the 1.6.0 branch?
1.6.0.25 is current, upgrade to there and then worry about the
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Tim Culhane wrote:
Here is my output of 'sip show peers'
user1/user110.41.3.12 D N 10434Unmonitored
user2/user210.41.3.12 D N 65293Unmonitored
user3/user3
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mir shahnawaz wrote:
Hi there,
I am running a PBX under asterisk 1.6. I have few FXO analogue lines
connecting to PSTN. These lines are in a hunt group. I trying to make
my extensions to dial 91, but this is a bit scary, I mean if somebody
make
UIT DEVELOPMENT wrote:
Sorry for what might seem as really silly questions, but I am not sure
how to proceed.
Thanks in advance for any insight that you folks can provide!
Hello Mike.
Welcome to the wonderful world of Asterisk. Before you sludge through a
GUI and all the attendant bad
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Ben Schorr wrote:
I’ve got G.729 loaded in the modules on the Asterisk server and on the
Polycom phones I’ve set G.729 to be the first preference of codec, but
still when I go SIP SHOW CHANNELS during active calls it still shows
“(ULAW)” (G.711)
Daniel Stefanus wrote:
Hi,
I want to reconfigure my asterisk dialplan.I have a problem.I have 4
agents in a queue.How is the configuration for the asterisk dialplan if
I want to have only 4 agents maximum who can receive the phone,so if the
fifth caller try to entering the queue they will be
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David Gibbons wrote:
This may belong on -biz, but does anyone have experience with a decent and
cheap IVR/prompt recording house?
Are decent and cheap mutually exclusive?
A nice *sounding* lady would be nice... you can keep any burly voice
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jonas kellens wrote:
Is it possible to make use of queues for incoming calls but to have
agents that do not need to log in ?
Make the phones members of the queue. In queues.conf:
[MY_QUEUE]
member = SIP/1234
member = SIP/5678
etc.
Barry
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B.Masoud @ SH wrote:
How can I set a maximum call duration on a ZAP channel?
Look at the parameters on the Dial application.
Barry
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James Texter wrote:
I found that on a clean boot, I could not connect to Postgresql either.
In my /etc/rc.local, I unload cdr_pgsql.so, sleep 15, then reload the
module, and that seems to work. After bootup, cdr_pgsql.so is able to
connect immediately.
This sounds as though you have
Karl Fife wrote:
Perhaps there's an arcane way to query lipbri the older releases from the
CLI? Can anyone speak to that?
Quick and dirty:
strings /usr/lib/libpri.so
That's CLI, tho' not the one you're talking about.
Barry
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Leif Madsen wrote:
Please review and let me know how it goes for you!
Where is it?
Barry
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aster...@opensourcesolution.in wrote:
Hi,
hi all.Iam new to VOIP so plz forgive me on asking stupid questions. I
have installed Asterisk on Centos 5.3,
and dowloaded X-LITE softphone on two windows machine. now i want to
start from very basic scenario, i want to make two X-Lite phones
velusamy velu wrote:
Dear All,
I have installed Asterisk 1.6.1.9 to use Bridge Application in
AMI. After inatallation I have tried to connect the AMI via telnet. But
it didn't connected. I used netstat to know the listening socket. But
it was not available. How to start the AMI server
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Darrick Hartman wrote:
I don't think that Maildir or a database backend solution (such as
Exchange) suffers from this same limitation.
Maildir makes sense, but the text I quoted in an earlier message is now
no longer part of the imapstorage text.
Kevin P. Fleming wrote:
It's not present in the current 1.4 doc/imapstorage.txt file, or any
later version. I don't even know why the storage format would matter,
since that would be very specific to the IMAP server that is managing
that folder.
Hmmm
Randy R wrote:
I missed the first part of this, but has anyone said: not all the
presentations were recorded.
Hi Randy.
Yes, that was mentioned. Actually, three of the four tracks were
videotaped IIRC.
Barry
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Darrick Hartman wrote:
It would be great if you could make more of the talks available to those
that attended the conference. I know there were a few times where two
interesting talks happened at the same time.
I have to agree John, I'd love to see the videos of the sessions that I
missed.
Kevin P. Fleming wrote:
That's a bug; the IMAP folder prefix should not be used in construction
of prompt names to be played back. Please open an issue on
issues.asterisk.org reporting this problem ...
Done: https://issues.asterisk.org/view.php?id=16104
On a side note Kevin, my INBOX is in
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I am running 1.6.0.15 and am trying to get IMAP storage working. I
have had no trouble doing so, except that I wish to create a subfolder
in my account for voicemail, such that I have:
#voicemail/
INBOX
Old
Family
Friends
Work
I can
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Barry L. Kline wrote:
I know that it is bad form to reply to yourself, but here is the current
state of affairs:
file.c:950 ast_streamfile: Unable to open vm-#voicemail.INBOX (format
0x4 (ulaw)): No such file or directory
I'm thinking
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John A. Sullivan III wrote:
I can't help you directly but I can share my experience with folders. I
intentionally did not set up the folder structure in IMAP as recommended
in the documentation. To my pleasant surprise, when the folders were
John A. Sullivan III wrote:
I'm using the INBOX - John
I'm throwing in the towel and going that route as well. Now that I'm
not trying to swim upstream things are working well.
Thanks John.
Barry
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James Lamanna wrote:
This is with dahdi 2.2.0 and asterisk 1.6.0.10.
Any ideas on this issue?
Check to see if this is a bug that has been fixed in 1.6.0.10. I
think the current is 1.6.0.15 and there has been significant bug fixes
since your
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C F wrote:
I have seen lots of companies offering this as a service and have used
phonetag.com in the past.
They work very nicely, however I have a customer that is not
interested in paying $30-$40 a month but would rather buy the
software. I
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ABBAS SHAKEEL wrote:
Can i know which querry is executed to insert record to database...
i am asking this because of
[Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:309 pgsql_log: Failed to
insert call detail record into database!
[Sep 1
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ABBAS SHAKEEL wrote:
but when i execute this ./configure --with-postgres=dir where
postgresql is installed
it gives an error for missing an pg_config file . i searched the PC
but it really dont exists. but database server is fine and
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laurent schweizer wrote:
how can I indicate to asterisk that number of message has changed and
that he need to do an update.
I'm not sure what effect realtime DB has on it but did you notice the
voicemail.conf parameters: pollmailboxes pollfreq ?
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Andy Kuo wrote:
Hi list,
I'd like to have the callers to listen to the advertisement (music on
hold) before the agents answer them. So, I have wrapuptime=10 in
queue.conf, but the call still goes straight to the agents without
delay.
Andy
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Andy Kuo wrote:
Hi Barry,
Thank you for the hint, but I forgot to mention that we have a few
advertisements, and we want the callers to listen to only one at a
time, and in a round robin or random order. Using Playback() doesn't
seem to serve
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Mauro Sergio Ferreira Brasil wrote:
We are planning to use Asterisk on our VoIP platform, and we are
spending some brains on a way to provide the following facility: let
some SIP user (extension) registrate with more than one client (ATA,
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Mauro Sergio Ferreira Brasil wrote:
I totally agree with you that this is an unnatural behavior, but I have
to agree as well with our commercial staff because their vision was
naturaly translated from our telephony world (we don't have a
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Thalassoline - Service technique wrote:
I want to show the QueueName to my Queue Member.
I try to find the solution to show the QueueName near the callerid of
each call.
Can somebody help me ?
Set the CALLERID(name) variable prior to sending
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Dáibhéad Antoine O'Reilligh wrote:
Have I forgotten anything?
Do you have 'sox' installed on your asterisk box?
Barry
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I tried patching my 1.6.0.10 source to 1.6.0.12 and during compilation
(make) I get a string of errors in chan_sip.c that start:
chan_sip.c: In function \u2018handle_incoming\u2019:
chan_sip.c:18669: error: expected expression before \u2018\u2019 token
chan_sip.c:18674: error: \u2018ret\u2019
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J. G. wrote:
I've been Googling all morning and searching voip-info.org
http://voip-info.org but not quite finding what I'm looking for.
I've read that you can modify the billing/account information on a CDR
via AGI but I can't find an example or
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Tarek Sawah wrote:
First of all it acts like a firewall and a router.. compared to Cisco
routers it has good ACL and firewall policies that can be used and
written very well..
second it's easy to setup
third my question is has anyone tested it
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Darrick Hartman wrote:
upgrade-run-image check http://mirror.astlinux.org/firmrware
Note the typo: firmrware
The working command is:
upgrade-run-image check http://mirror.astlinux.org/firmware
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I'm trying to install my first channel of FAX. When I run the benchfax
utility, I get to various stages and then the program simply hangs.
There is no excessive CPU utilization and the benchfax program readily
responds to ^C. Sometimes I get fairly far into the test, other times
it hangs almost
Kevin P. Fleming wrote:
You say it has stopped at different points as you've run different
tests; can you verify that? If it always stops at 'nocona', that would
mean one particular type of problem, but if it stops at various
different places, then that would point to something very
Kevin P. Fleming wrote:
If you can, run it again under 'strace', capture the output, compress it
with gzip or bzip2 and email it to me directly; that may give us a clue
what it was doing.
Requested strace output sent.
Barry
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Trevor Hammonds wrote:
I am hoping someone on the list has an example of a lightweight AGI script
that I may modify to either read the simple text file and set a dialplan
variable to the current temperature, or hopefully a more-sophisticated one
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Mark Michelson wrote:
Thanks for the config info. I have a couple of suggestions for fixes.
1. Try changing the type in [basic-options] from friend to peer. I've found
that
device state reporting for outbound calls (from the perspective of
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Danny Nicholas wrote:
Hi Gang,
Running Asterisk 1.4SVN using Polycom 501 phones. Just
enabled CallerID and for the most part it works as good as you’d expect
anything to from the phone company to. Except: on about 1 out of 10
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I must be missing something here but I can't figure out why I can't get
DEVICE_STATE() to give me anything other than NOT_INUSE.
I have two extensions: and 6668. I used 6668 to make a call to
yet another phone, so I know that it's busy. I
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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, but no busylevel set. These
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Philipp Kempgen wrote:
Just to be sure: Do you have hints configured for the extensions?
See http://das-asterisk-buch.de/2.1/blf-leds.html
(The text is in german but there are many examples in extensions.conf
and extensions.ael syntax. Zurück =
John A. Sullivan III wrote:
Hello, all. I'm having a nasty problem with call parking in Asterisk
1.6.1.1 that smells like a bug. When the call returns, it seems to be
returning to a | delimited extension and failing. Here is the output
from the console:
Hi John.
I've just run into the
Jonathan Thurman wrote:
This was fixed in the 1.6.1 SVN, and I would guess that it was also
fixed in the 1.6.0.
SVN log:
r189951 | russell | 2009-04-22 11:56:43 -0500 (Wed, 22 Apr 2009) | 2 lines
Fix call parking callback. Pipes - Commas.
You will have to create a patch against
Barry L. Kline wrote:
I'll figure out how to make this patch
against 1.6.0.10.
That was a trivial fix. I hope that they permanently add that patch to
the 1.6.0.x series.
Thanks again Jonathan.
Barry
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Jeremy Winder wrote:
I'm in the process of converting our current hybrid key system to
Asterisk and Aastra 57i phones. One of the features that seems to be a
show stopper for almost everyone in the office is the inability to see
who is on the
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Barry L. Kline wrote:
If you'd like a more generalized approach you can install an Openfile
server and use the Asterisk plugin. That'll give you an internal IM
server which will show the status you seek.
Sorry, not 'openfile' but 'openfire
Kevin P. Fleming wrote:
There is no need; your existing Cepstral-supplied licenses will continue
to operate, and will be added to any Digium-supplied licenses you
purchase and activate.
Hi Kevin.
That didn't work. If I use 'swift -n Allison-8kHz -o test.wav Hello,
test and play the
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I have a license for Allison-8kHz and two concurrent port licenses that
I purchased from Cepstral at the end of last year. I just got around to
installing to my * 1.6.0.10 machine.
I've decided that the best way for me to integrate the two would be
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Kevin P. Fleming wrote:
There is no need; your existing Cepstral-supplied licenses will continue
to operate, and will be added to any Digium-supplied licenses you
purchase and activate.
Thanks Kevin.
So I shouldn't worry about this?
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Karl Fife wrote:
I discovered that after running make, you can run 'make sounds' before
shutting down the service. This cuts all of the download time from the
install process minimizing service downtime to a fraction of what it would
othewise
Clara Chan wrote:
Loan,
Thanks for your help in this matter.
Having never used astdb before, can you point me to an example on this??
Thanks hugely,
Clara
Clara --
You need to read the book. In it you'll find examples.
Asterisk: The Future of Telephony 2nd Edition (ISBN
sean darcy wrote:
Maybe I've not explained this correctly. I know, or can look up, the 40+
local exchanges that are local. I can parse the dial EXTEN to determine
the exchange. I can check the exchange against a DB. I want to determine
which exchanges are local. I do not want to store an
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David Backeberg wrote:
5) exten = s,n,Dial(${OUTGOING_PRI}/${ANSWERINGSVC},15,r);
6) exten = s,n,Goto(s-${DIALSTATUS},1);
What is the 6 for?
What is the goto supposed to do?
Hi David.
The '6' is in case I get a CHANUNAVAIL or other error back
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sean darcy wrote:
I've looked at the Berkeley DB. That works pretty well, if the exchanges
are all stored. But it looks like the exchanges have to be entered 1 by
1 from the CLI. And can only be reviewed, corrected, or deleted from the
CLI. I
Danny Nicholas wrote:
You should try Answer before Dial on the Monitored call. Bridging can be
very unhappy.
Hi Danny.
Already done earlier in the dial plan, when the call first comes in but
before it gets routed to the part that I showed. Thanks for looking
though!
Barry
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Robin Rodriguez wrote:
still rather frustrating getting the EFK working. If needed I could
post that portion of sip.cfg to get you started.
Please do! Just having the example could be helpful for those of us
preparing to tackle this kind of
Danny Nicholas wrote:
To clarify:
Inbound - Answer
Outbound - Answer (again)
Dial.
Hmmm... that seems like it would be from the department of redundancy
department but I gave it a try, both before and after the Monitor()
command with the same result... it fails.
Thanks!
Barry
I wrote a note earlier about this problem but have done quite a bit more
debugging. Now I'm stuck at what to do next.
I have inbound calls being answered by our Asterisk box, which then
dials our answering service and bridges those calls. The inbound and
outbound are both PRIs. The answering
This is getting really interesting. I had a chance to do some testing
last night. To recap, here is what I'm attempting to do:
Caller -- INBOUND_PRI -- Asterisk -- OUTBOUND_PRI -- AnswerService
The caller dials our number, * picks the call and offers some choices.
If the caller needs to speak
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Sorry for the delayed response, I was out of the office.
David Backeberg wrote:
But there's not the native bridging status on the calls with recording
enabled, where as the native bridging report fires on the
recording-less dialplan.
A clue
I have an application where we receive calls on an inbound PRI. After
hours, our Asterisk box dials our answering service on an outbound PRI
and then bridges the caller to the answering service. The flow looks
like this:
(CALLER)INBOUND_PRI -- CONTEXT -- GOSUB(Incoming) --
David Backeberg wrote:
I don't know why recording is breaking your calls. My guess is
something is screwed up with your PRI configuration. Are you getting
alarms in your logs from dahdi?
Not a peep, either with or without using the monitor command. I've
been using this system for around
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Dean Collins wrote:
Technoco vdex40 is probably outside of your pricepoint but you might
want to consider them
My experience with the vxex40 was not great. This was about six months
ago. I'd not recommend one unless I could ensure that I got a
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Jeff LaCoursiere wrote:
What were the issues?
There were tons of issues involving the generated configurations (such
as not being able to use attended or unattended transfers) but the worst
issue was that the box would not hang up POTS lines. We
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Andrew Nowrot wrote:
This had happened twice so far. Does anyone know what is causing this.?
Start by upgrading to 1.6.0.9, then if it continues you can start
tracking it down.
Barry
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I remember someone wrote a great document concerning Polycom server
provisioning that provided a way to ensure that updates to the firmware
did not overwrite customizations. I'll be damned if I can remember
where I saw it. It may have been
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Kevin P. Fleming wrote:
It's easy; just don't edit the files that come with the firmware!
Hi Kevin.
That's the model I currently use. The one I'm interested in is linked
in Darrick's post below. It's an interesting approach.
Thanks for
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Darrick Hartman (lists) wrote:
That would be Karl Fife, of the famous Karl Fife experience.
http://kfife.com/voip/
That's what I'm looking for. Thanks Darrick!
Barry
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jonas kellens wrote:
I have 2 questions about the following output on the Asterisk CLI :
Jonas
Please do not hijack threads. Please start a new message. There is a
good chance that many people didn't read your message because it's in
reply to another, unrelated one.
Barry
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Sriram wrote:
1. I need to store the CallerId of the PSTN caller with his language
preference so that next time he is played the prompt in his language
that he chose the first time.What would be better - storing his number
in the Asterisk DB and
--[ UxBoD ]-- wrote:
I am going to be building a new home Asterisk server this weekend
(Dual core Intel Atom 2GB RAM) and would like to ask whether it
would be worth starting fresh with a 1.6 install instead of the 1.4
one I have at the moment ? I do not have a complicated dialplan as it
Adrien Lemoine wrote:
Maybe someone experienced something similar and can drive me in the
resolution ?
You have given no information about your hardware, OS, Asterisk version
or what you need to do to recover the system (e.g. reboot, just restart
Asterisk, etc) so no one is going to be able to
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jonas kellens wrote:
I pick up the phone, and dial 211 on the BT201. This is the Asterisk CLI :
/Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3895)/
/Verbosity is at least 5/
/asterisk*CLI /
Nothing is displayed... it
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markus wrote:
Hello everyone!
I installed Asterisk following the instructions of the book
Asterisk: The Future of Telephony. (very nice book)
However, I failed.
I installed zaptel, libpri and asterisk (in this order).
If you are using
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Ken D'Ambrosio wrote:
So: what/how do I need to install to meet this dependency?
Did you run configure again after installing the missing components?
Barry
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Brandon B. wrote:
At the top of my /etc/dahdi/system.conf file is this line:
# Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10
2009 -- do not hand edit
OK, so how do I adjust the timing source and LBO numbers, and echo
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Jared Smith wrote:
While I personally believe it's a bug, it has been in Asterisk for a
very long time, and I know from teaching Asterisk training classes that
there are *many* *many* people abusing this in their dialplans. I'd be
quite hesitant
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David @ULC wrote:
When I am trying to delete voice logs,
[r...@vicidialnow monitor]# rm * -r -f
-bash: /bin/rm: Argument list too long
[r...@vicidialnow monitor]#
Argument list too long is coming as a road block.
Now way to forcefully
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Here's one that may be of interest to any upgraders. If you rely on the
behavior of gosub you may want to make note of this change.
I have an incoming call context:
exten = _,n,GoSub(incoming,${EXTEN},1(${EXTEN}));
that is supposed to gosub
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David Ruggles wrote:
global variables that link the cell phone #'s and extensions and have this
done somewhat automagically.
Load your cross-reference in AstDB and do the lookup that way. If the
cell number exists in the database, replace the
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Steve Gladden wrote:
Is 1.6 so cutting edge that I should not expect to find complete
documentation (yet)like I seem to be expecting very easily?
Most of what is applicable to 1.4 is applicable to 1.6. I'm running 1.6
without any hiccups -- YMMV.
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Mark Michelson wrote:
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 --
This bug still exists in the recently-released
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Tilghman Lesher wrote:
Mark erroneously assumed there would be another release candidate, which
there was not. So while it's not in 1.6.0.3, it will be in the 1.6.0.4
release, when that occurs.
Thanks Tilghman. I wait with bated breath.
Best
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Mikel Lindsaar wrote:
What does putting ww at the front do?
Each w makes Asterisk wait a 1/2 second before sending the DTMF to dial.
(It may be a 1/4 second each 'w')
Barry
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Bill Andersen wrote:
In the order in which people normally read text they don't
repeat the entire conversation from the beginning each time
a question is asked either... Bottom posting is just as bad!
./bill
Not when you take the time to properly trim your reply it's not.
BK
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