Here's a good example. I'm trying to get SIP blf. I managed to split
my
result into a list of lines by splitting on ANY of \r\n, \n or \r. I
was
going use the column headings from the third line as my keys for my
dictionary/hash, rather than hard coding them. Notice anything? The
'Call
ID'
Mitch,
I was told that 'top' can be misleading because of the fact that Linux
will generally use free space and allocate it to cache. Try running
your tests with the 'free' command and see if there really is a memory
leak. I think the real concern is the fact that calls start dropping
once
I'm interested in knowing if anyone else has worked around this issue:
I have an application that needs to check the status of the calls going
through Asterisk about every 5 seconds or so. I don't want to do
asterisk -rx 'show channels verbose' at the Linux command line 12
times per minute so
you need to look again
maybe read, http://www.dynx.net/ASTERISK/DOCS/RTF/MANAGER.RTF
Mea culpa!
I did not realize it was so easy! I totally missed the command action
when looking over the docs. Thanks for pointing out the mistake and
thanks too for the link. It has helped a lot.
-MC
It wouldn't be hard to code up at all actually... a little perl magic
and
voila. ;)
Who needs a weekend project?
The Perl magic would be easy. Writing the check to pay for all of that
data is what is so hard...
-MC
___
--Bandwidth and Colocation
Trixbox scatters it's config files. Some stuff is kept in the
database, some in the conf files.
You have to keep your configuration in specific files that won't be
overrritten.
True - TB does a lot of very specific stuff. If you want to have a
plain Jane dial plan for your stuff then use the
Gang,
I'm having this error pop up when I do a ForkCDR, and I'm not sure how
to get around it. Here are a few log lines:
Nov 8 10:37:08 VERBOSE[28079] logger.c: -- Executing
ForkCDR(Zap/49-1, ) in new stack
Nov 8 10:37:08 WARNING[28079] app_forkcdr.c: Channel does not have a
CDR
The
I have a debugging scenario where I wish to record the entire call. The
call is establish via a .call file. I can't seem to get Monitor to do
anything. My dialplan looks like this:
[dialout]
exten = s,1,DigitTimeout,1
exten = s,n,ResponseTimeout,10
exten = s,n,Answer
exten =
We've been using DUNDi, Realtime, and regexten extensively for months
now, and it's been working great since we got it running.
Could you please tell us a little about the experiences you had in
getting it running? Evidently there's some magic involved, otherwise so
many wouldn't be struggling
I first learned asterisk via [EMAIL PROTECTED]
Then I went to straight asterisk.
This seems to be a theme. Getting your feet wet with [EMAIL PROTECTED]/Trixbox
is not
a bad way to go, especially if you want to get a functioning system up
and running quickly. After tinkering with
I'm not siding with anybody here, but there is some glaring
mis-information in this thread.
-Matt
Matt,
Thanks for the information and the Wikipedia reference. We all
appreciate unbiased presentation of facts, even if we don't always
present them in an unbiased manner ourselves. :)
-MC
Exactly. Keeping some extra TDM hardware around for several customers
and
keeping the configurations when the drive dies can do wonders for your
guru factor. It's common sense, really. What's your cost for
keeping a box around for development and running it out to a
customer in a crisis
Yep or even less these days with the increases in technology.
Having a hot spare doing nothing is cheap insurance.
Cheers,
Dean
*VERY* cheap insurance, especially when you consider how devastating
downtime could be!
___
--Bandwidth and
After working with NEC systems for more than 10 years, both as a
technician and as an end user, I can say with confidence that their
stuff just doesn't break. Period. You can kill it by installing it in
an unventilated phone closet, outside and exposed to 110F degree Fresno
summers, but even
How do I use priority n correct?
First, which version of * are you using? Hopefully something recent.
If you've got 1.2.x then you can use n and labels. Check this out:
http://www.voip-info.org/wiki/index.php?page=Asterisk+priorities
Tinker with it - you'll be surprised at how easy it
The calling side needs to be PRI since that is the side that
receives the information. The called side doesnt matter the called
side could be analog, but if the calling side has PRI then the calling side
still receives a cause code. The problem is that many telcos dont play
nice and
Dude,
Enough with the sales pitches already! Just provide a link and say, You can
get Italian prompts here, and be done with it. Either that or learn the
difference between the users list and the biz list.
-MC
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
kissing up some more
FYI, you both are right! Getting started with AAH/Trixbox can be very
valuable, but relying upon it can be very limiting.
/kissing up some more
I started w/ AAH, then went back and learned the dialplan apps,
scripting, etc. For some guys like me, it's easier to start with a
My question to everyone is this..This is where I am
at now. I have been using FreePBX for about a year, after moving from [EMAIL PROTECTED] I am
starting to need some manual changes and modules. My question is can anyone
point me in a direction on how to learn how to create these. I
Asterisk is what you make of it. If you don't want certain
applications to run on a certain instance/machine then you should
noload them in modules.conf.
Barzilai still has a point. Noloading various applications doesn't
address the underlying architectural issues. The fact of the matter
Douglas has a point, and a legitimate one at that. Setting aside your
personal feelings about Doug and his style of commenting, please
consider that 'lack of documentation' is either the first- or
second-most-often cited criticism leveled against open-source software
and the OSS community. Lack
Doug,
I'm sure that you are not the only one who considers an API w/o docs to
be of limited or no value. I just doubt that many people have use for a
management API because they don't use the Asterisk manager interface
very much.
On a side note: some folks who have limited programming
I guess I simply have a different viewpoint. If I document, even
lightly, something that I give to the general public (or a specific
group within that general public), I feel I *WILL* get a ROI. The more
people who use my 'gift' the better. Who knows if someone will return
to me a snippet of
Hi, can someone enlighten me as to the difference between a PRI and a
Digital Trunk (other than cost)?
Barry,
A digital trunk from the telco is most likely as you said - 24
channels of 64k voice. (I know, technically it's data but the data
is just digitized voice.) Think of the digital trunk
There needs to be a *CLEAR* policy on when spaces are and are not
stripped!
This type of bug would be murder to track down! Spaces are typically
stripped from configuration files, and to have ONE variable type in
the
config file behave differently is an *unbelievably* poor design.
Was
Bruce,
Good call on this one! Ive found that
users can handle small changes if they are parallel with something theyre
already comfortable doing.
-MC
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Tuesday, July 18, 2006 1:29
PM
To:
Lincoln,
Check it out, this is your friend:
http://www.voip-info.org/wiki/view/Asterisk+PRI
This will help you get your zaptel.conf and zapata.conf settings all in
order.
-MC
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Lincoln
How about this app:
NoCDR()
I.e.
exten = s,n,NoCDR()
-MC
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt Gibson
Sent: Thursday, July 06, 2006 1:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
you say Flash asterisk command send a flash signal to old pbx so that
it
sees that command as coming from an analog phone. But since Flash is
not
a digit, how can I catch it from within asterisk? How can I tell
asterisk (es inside extensions.conf) to do something whene receive it
from a
I get annoyed Stephen when Digium goes around calling Asterisk
'enterprise grade', which in my opinion it really isn't. I'd consider
distributed ACD queues to be a requirement for an enterprise grade
product, but it's becoming apparent that there is no mechanism for
implementing this. I'm
I'm curious what a manner of speaking
is. If I go that route what am I losing? I really just want to make
sure whichever route I go I will be able to come here for help and not get
blown off because of something non-standard in the packaging I chose.
By manner of speaking I mean
Has anyone created a script that will download and install all of the
freepbx prerequisites in the INSTALL file automatically on a Centos 4
box?
In a manner of speaking the trixbox guys have. Have you ever seen that
(or Asterisk @ Home)? There is a script, install.sh, that installs a
bunch
Al,
Are you doing voice broadcasting
that is, delivering a pre-recorded message, possibly giving a live caller other
options? Just curious. Ive been working on a
voice-broadcasting application myself and Ive had mixed success with
app_amd.c. It does work very well in some cases, but
Hi list!
I have a Centos 4.3 box running Asterisk 1.2.9.1 with FreePBX 2.0.1
I noticed that this setup is keeping a full asterisk log which, after
1
month in production, has already grown to 1300 Mb in size. This is the
log
location : /var/log/asterisk/full
Why is this on by default (I
Kevin P. Fleming wrote:
According the Sangoma data sheet, the Octasic part _is_ the DSP (which
it
is, in a logical sense). The board does not relieve Asterisk/Zaptel of
any
additional burden beyond echo cancellation and tone detection at this
time; Asterisk/Zaptel don't know how to take
But the high dollars don't generally get you the high processing
power,
or a solid quality product (cough, Dialogic, cough).
Agreed. It's another case of perception vs. reality. Having some
processing power on the card is always better than none - or so many
vendors would have us believe.
If you can't afford to purchase both cards, then a safe bet is to
simply
purchase the Sangoma card since it can address more echo issues then
the
Digium card.
Also, don't forget that the high-end A104d has more than on-board EC.
It has on-board DSP handling and a 5 year warranty. Check it
The installation of CentOS is sufficient to support TrixBox, but you
can
always add additional packages using yum.
If you download the iso and view it then you can see which packages
Trixbox loads by default. On the CD or in the iso file find this
directory:
/CentOS/RPMS/
In it are all of
I use the goto to jump across contexts with labels all the time.
goto(context,exten,label). works for me.
Cool. If you have some time, tell the developer to update the docs!
Definitely cool. I will test this out and if I can make it work on my
test machine then I will be happy to update
If PRI channels are not dedicated to DID number - that's all I need. I
hope it will work as you describe and there will be new call on new
channel each time they dial same number. That's VERY cool.
Sorry, I am a newbie to T1 world, though have been using Asterisk with
TDM400P PSTN lines /
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Goto
show application goto
-= Info about application 'Goto' =-
[Synopsis]
Jump to a particular priority, extension, or context
[Description]
Goto([[context|]extension|]priority): This application will cause
the
calling
Regarding my earlier post about labels and the 'n' priority:
The TFOT book covers the use of these. See the box on page 81 entitled
Unnumbered Priorities.
http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip
-MC
___
--Bandwidth and Colocation
Thanks Michael. I was not aware that labels where available.
In converting though, I've already hit a limitation. There's a single
name
space for all labels I assume?
Doug,
According to TFOT's Goto() application reference entry (page 254) the
namespace is actually the current extension:
Named
Anyway... How can I use Goto() to jump to a label in a different
extension
or context?
When you have a lot of loops and such in a single extension, you end
up
wanting to use multiple labels called 'start', 'next' etc. I
assume(hope!)
that the namespace of labels is in a single context? ie
Oh Crud. So, if I want to jump to another extension or context, I have
to
specify the full context, extension and priority? I can't specify a
label?
It's a bit tricky trying to jump to a specific priority in an
extension
when they're all called 'n' !
Why is something so simple such a
Do you have a before-and-after example? I
think wed like to see a sample of a context extensions with
hard-coded priorities and the subsequent translation into unnumbered priorities
with labels. There are some creative people out there who might have the key
to getting your dialplan
I apologize for my silly prior response, I didn't read the thread
enough
:(
Your humility is much appreciated!!
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Is it going to be sticking around? I sure hope so, because the
extensions.conf language is a management nightmare. Every time you add
logic, you have to renumber everything, and hope you don't make
mistakes
in re-setting all the dependant goto's. It ain't exactly easy to read
later on either.
How would goto work if all the priorities where n?
...
Example from one of my dial plans:
exten = talk,1,ForkCDR
exten = talk,n,Set(NUMTRIES=1)
exten = talk,n,GotoIf($[${NUMTRIES} = 1]?first)
exten = talk,n(repeat),Background(Initial-greeting)
exten =
you should mv the file (and in the same filesystem, so 'rename' is
used)
You might want to chmod or even chown the file first as well. I wrote a
little script that does all of this before the .call file is mv'd into
the outgoing directory:
cp /tmp/test3.call /tmp/test1.call
chmod 666
For those of you who saw my gargantuan post the other day I'd just like
to say thanks for listening and sorry for the lengthy post!
It turns out that my key issue was with the WaitExten app. I saw this
on the wiki which really helped out:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
I would appreciate hearing from anyone who has figured this
one out. Heres the scenario:
I have a context wherein I give the called party the option
to dial the digit 9. If he does so, he is transferred a la this extension
entry:
exten = 9,1,Playback(pls-hold-while-try)
exten =
I've never bothered to check to see if cat5 cables use the appropriate
mating twisted pairs or not. Since the pinouts are different for cat5
vs
T1 cables, I'd have to guess a single strand is used from two
different
twisted pair groups. That wouldn't be cool, but in short runs it
probably
Close. 10/100mbps Ethernet uses wires 1,2,3,6 but that is pair 2 3.
Pair
one is the pair up the dead center (pins 45), pair 2 is pins 12,
pair 3
is
36 and pair 4 is 78. A T1 uses pairs 12, which is why you can't
use a
regular crossover cable for a T1 crossover, but you can use a regular
A few months ago I needed some help for the following issue:
.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the
call
.) the caller get lost at this point !!
At this point the
Jerry Jones wrote:
Yes it should all behave the way we are used to. However SIP IS
different. The exact behavior will be dependant upon the individual
hard phone.
Isn't that true only if it has a preprogrammed transfer key?
an Asterisk feature code should work as discussed.
There
show application dial Pay special attention to the D() option.
Eric,
Question - does the D option know that on a PRI the DTMF stream goes out
the B channel and not the D channel? I would assume that it knows but I
thought it best to ask the question outright. If it does, then it
should be
That's way too much Java for me. I'm lost already.
Doug,
I'm a Perl guy myself, so I think in terms of Perl and CPAN. I'm sure
Python has its own version of CPAN where people upload modules for other
programmers to use. CPAN has a Perl module:
POE::Component::Client::Asterisk::Manager
It's
the DIALSTRING you were given is just an extension, 089324154332. As
Lenz pointed out, and it also says in the app_dial.c:773 WARNING, it
must be technology/number, not just a number. Not sure perl methods,
but you might concatenate a technology before the number, something
like
$res =
I'm using frequently the perl api within asterisk.
Now I'm looking for documentation for the perl commands.
Some perl commands I found on this URL:
http://www.voip-info.org/wiki/view/Asterisk+PHP
Does anybody got more documentation or where I can found some more
documentation about perl
Forgive me if this question has been asked/answered in another post.
And let me reiterate what other users have frequently said - Asterisk
is
great, and I really appreciate all the work you folks have put into
it.
How have some of you gone about integrating Asterisk with a legacy
office
Does zapata.conf have any function in systems that aren't using
zaptel(
I suppose not)?
Just curious - what driver are you using? (I'm not familiar with
wellgate.)
I am using an external gateway (wellgate 3701a) and don't have zaptel
at all.
If I am not using zapata.conf (this is my
There's a book on my desk right now that disagrees with you...
ISBN: 0-596-00962-3
I believe Doug's experience with the TFOT book's DUNDi section was less
than stellar. If memory serves, it is possible that some of the
examples from the book were out of date. A few months back there was a
Hey all,
I have a situation where I have 8 lines from the phone company in a
hunt
group coming in to my asterisk box. These are the same lines I'm
using
for outgoing calls ( named g0 ).
The problem arises when someone dials our number at the same time
asterisk tries to put a call out on
For the record, Douglas is correct on this point of enterprise-grade
being on ABE:
http://www.digium.com/index.php?menu=product_categorycategory=software
Copied and pasted right from the website, it says:
Asterisk Business Edition(tm)
Digium(tm), the leader in open source telephony, offers
I actually got it all working - but it's great to see where we did the
same
thing, and where we differ.
I ended up using the 'pop' perl command - inside a loop to go back one
item
at a time through my list
PaulH
Nice work! Perl = TMTOWTDI = There's More Than One Way To Do It
-MC
Thanks for this example - it has really got me started!
Paul,
I did some tinkering and I think I found something that might be
helpful. If not, I did at least learn quite a bit about AGI scripting
and dialplan writing! :)
Okay, first I created a pretend file with numbers: /tmp/numbers.txt
Paul,
Just curious - what kind of stuff are you reading from the file?
-MC
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, February 27, 2006 7:53 PM
To: Asterisk Users Mailing List - Non-Commercial
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, February 27, 2006 7:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Asterisk Question
I was going
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, February 27, 2006 7:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Asterisk Question
I was going
That's getting pretty close - thanks for that.
I just couldn't find any decent info on the web about working with
AGI.
Ditto. However, I pieced some stuff together by sifting through my
well-worn copy of TFOT and bouncing around between the wiki, the sample
AGI scripts and
Thanks for this example - it has really got me started!
Short question - how can I put a variable into my perl script?
I imagine it's something like
exten = 780,1,AGI(agi_ret_val2.pl|${back})
But how can I get my perl script to pick this value up?
Again - thanks to everyone who has
I'd like to use the AGI command CHANNEL STATUS to check the status
of a
channel. However, the dial() command doesn't return -1 until after the
call has hung up. If that's the case, how is channel status supposed
to
return statuses like:
status values:
0 Channel is down and available
1
Douglas Garstang schrieb:
...
HOWEVER, if the CALLER hangs up the call, it seems
Hi,
did you try the dial command option g?
I did not neither, but when I understand the voip-wiki right,
it might help you.
Roger.
I've used the 'g' option and as far as I can tell it works just
MC,
I think I worked out that I need to use ${DIALSTATUS} anyway. Don't
really
see what 'channel status' is for...
Doug,
I think the channel status might be more useful for incoming calls, or
possibly to monitor specific channels other than the 'current' channel.
I can't think of a reason
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DeadAGI
Cheers,
Jean-Michel.
I am red-faced! The TFOT book explicitly says this on page 158, on the
box titled, AGI(), EAGI(), DeadAGI(), and FastAGI():
The DeadAGI() application is also just like AGI(), but it works
correctly on
If that's true, why does dial() return control to the script when the
callee hangs up?
Doug, if I understand the AGI limitation correctly, the 'dead' in
DeadAGI() refers to the other end of a dial() connection. I *think*,
but I'm not positive on that.
Does anyone know the answer to this
Hi everybody,
This question is confusing me for some time. From selling point of
view
to a customer, calling asterisk a PBX doesn't look right. According to
the definitions of PBX or PABX, Asterisk is not just PBX but much more
than that. My question is, how should I introduce Asterisk to a
Nitin Joshi wrote:
Hi All,
I have installed a Digium TE110P card on an Asterisk 1.2.1 system.
Its
connected directly to the PSTN. But I am unable to make outbound
calls
on the zap channels. The light on the card is green. Asterisk CLI
shows all 24 channels when I give the command 'zap
Dov Bigio wrote:
Hi,
I got this message on my Asterisk messages file and
after it Asterisk went down...
2006-02-18 08:13:55 WARNING[3214] ast_expr2.fl: ast_yyerror(): syntax error:
syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP
or TOKEN;
34
-Circuit/channelcongestion)
On 2/17/06, Michael Collins [EMAIL PROTECTED] wrote:
Nik,
This definitely helps! Please check your dial command. You've got
Dial(Zap/0/mynumber) and I think you might possibly want it to be
something like this:
Dial(Zap/1/mynumber) or
Dial(Zap/g0/mynumber
callsUnabletocreatechannelof type 'ZAP' (cause 34 -
Circuit/channelcongestion)
On 2/15/06, Michael Collins [EMAIL PROTECTED] wrote:
Nik,
Looks like you're making some progress. When I first started using
[EMAIL PROTECTED]
I had trouble getting the outbound dialing to work. I wasn't sure
where
FYI,
I found a workaround with this. Festival w/ [EMAIL PROTECTED] comes with 4
voices.
Here's a snippet from the siteinit.scm file:
;(set! voice_default 'voice_cmu_us_bdl_arctic_hts)
(set! voice_default 'voice_cmu_us_slt_arctic_hts)
;(set! voice_default 'voice_cmu_us_jmk_arctic_hts)
;(set!
Ken,
The zaptel.conf looks good as far as I can tell. The only question I
have is on the Zapata.conf - do you know for sure that the switchtype is
supposed to be national? Just curious. My telco's are all set for
4ess/5ess or dms100.
Second, can you do a debug pri span 1 from the CLI and
I can't. It was a one-shot deal, as (because of the phone company) I
can only get the T1 turned to ISDN during work hours, which means that
my company's lines are down while I'm trying to switch over. I
realized
about 1 minute after I told them to revert that I should have gotten a
debug pri
Just curious to know if anyone uses Festival with * and
whether or not youve got a different voice than the default. Im
looking at doing a commercial application but my boss doesnt want to
shell out the $ before we do some real world testing of * and Festival. Specifically,
Im looking for
/06, Michael Collins [EMAIL PROTECTED] wrote:
Nik,
I'm not sure that NOP is correct, but I'm in the states so I'll to
defer to someone who knows E1/PRI. When I run zttool I have OK
under
the alarms. Is there a way you can call the telco and confirm the
settings? Make sure that framing
Doug,
The TFOT book recommends using expressions instead of dialplan
functions:
exten = s,1,Set(mainLoop=$[${mainLoop} + 1])
-MC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Wednesday, February 15, 2006 2:10 PM
To: Asterisk
Yes - in a traditional PBX environment the transferring station has the
ability to pull the call back by pressing a sequence of keys. In some
PBX's, pressing the transfer key twice, like a double-click of a mouse,
will pull the call back. In some analog environments, pressing the
flash key twice
Nik,
Just curious - what is your telco setup? Do you have PRI with the
specified D channels? You need to make sure that your telco is set up
to have the D channels on 16 and 47. When you first start Asterisk, or
when you log on to the CLI, do you ever see messages stating the B
channels are
Subject: Re: [Asterisk-Users] problem with outgoing calls Unable
tocreatechannel of type 'ZAP' (cause 34 - Circuit/channel congestion)
On 2/13/06, Michael Collins [EMAIL PROTECTED] wrote:
Nik,
Just curious - what is your telco setup? Do you have PRI with the
specified D channels? You need
This can also be done with the use of call
files. Check this out:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
-MC
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arjan Kroon
Sent: Monday, February 13, 2006
7:10 AM
To:
Curious: Why did you need the wait times to be so long - was it because of your
PBX or is that simply what you wanted?
-MC
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hans-Juergen
Brand
Sent: Monday, February 13, 2006 2:12 PM
To: Asterisk Users
of type 'ZAP' (cause 34 - Circuit/channel
congestion)
On 2/13/06, nik600 [EMAIL PROTECTED] wrote:
On 2/13/06, Michael Collins [EMAIL PROTECTED] wrote:
When Asterisk first starts up, it will attempt to bring up the B
channels on any PRI circuits. If you are using [EMAIL PROTECTED] then you
can
JCC,
The issue boils down to this: how much work does the human have to do to
get the calls routed to the right place? In a traditional PBX
environment, a receptionist does not have to choose beforehand whether
he/she is going to do a blind or attended transfer. Like I said before,
how many
Nik,
Start here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels
It will give you some background info. Also, be sure to learn the
difference between zaptel.conf and Zapata.conf. It took me two weeks to
realize what each one does:
Zaptel.conf handles the lower-level stuff
John is absolutely correct - in the PBX world a transfer is a transfer,
regardless of whether it is blind or attended. How many PBX phones out
there have two different transfer buttons, one for blind and one for
attended? Zilch.
It's the user's behavior that determines whether or not the
FYI,
If you want to learn more about why ${EXTEN:1} works, check out the
Asterisk TFOT book, chapters 4 and 5. Page 95 of chapter 5 deals
specifically with the ${EXTEN} variable and the syntax of adding :1
(or :2, :3, etc.) - good stuff to know.
Check it out:
Perhaps there's a happy medium: sprintf()?
I am curious to know if putting the output into a char array with
sprintf() (to preserve the output formatting) and then writing it with
write(). How much additional overhead would this take? Hard to know
without trying it.
Is anyone in a position to
Kevin,
I agree with your assessment of the preference of using fprintf()
instead of sprintf() + write() + maybe malloc(). After hearing your
candid explanation it makes perfect sense not to pursue this. Ive
only been playing with * for two months, so Im still gathering my
bearings. As
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