I'm having a lot of problem with it recognizing oh for zero.
I've tried both o and oh. In one case, I just tried:
$digit = o { out = 0; } | fundamental {out = 2; };
So I gave it a choice that was VERY far away from what I said.
But when I said o o o o o, more than 75% of the time, it
I'm getting a crash relating to this field and I'm missing something.
It seems to be initialized to zero, then used in memmove, then
DECREMENTED. Where is it ever incremented?
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Is this bug alive in 1.6.2.9-rc1? I'm getting segfaults from chan_dahdi.
If it does effect 1.6.2.8-rc1, I'll just wait for rc2 to see if this
is my problem, instead of filing.
I reported another instance of this today and it was fixed in the SVN a few
hours later.
--
Seems to me a similar argument for and against hosting ones own web
presence in house with mixed results . Others choose to use a
datacenter service, seldom but sometimes with poor results.
I think that's a good analogy. It's very hard to argue that one of those
choices is right and the
I did a usual svn update, ./configure and make and got
[CC] chan_oss.c - chan_oss.o
gcc: @SDL_INCLUDE@: No such file or directory
I don't see any changes to chan_oss recently, so don't understand this.
What could be going on?
--
Suppose I have a subroutine (called by Gosub) S that's called from a macro
M and there's a goto to an illegal extension in S. That does go to 'i' in
S but seems to pop the macro stack so that when there's a later fallthrough
in M, the calls hangs up rather than returning to the caller of M.
Is
You really shouldn't be calling a Gosub routine from Macro. We've
already had to deal with some really odd interactions between the
two. If you're going to make the jump to Gosub, go completely over
and only use Gosub. Don't use Macro in conjunction with Gosub.
I find each most useful for
I'm working on my dial plan and I'd like to parse all dialed numbers
to convert them to the format I want.
For example if someone dials 0112345678 or 0033112345678 I would like
to convert it to +33112345678 and then match the number to my exten =
+33 statements.
If I understand what
I am looking for a voice recognition technology integrated to
asterisk. Any suggestion about it?
I'm using the Vestec product from Digium and having good luck with it.
There's also LumenVox from them as well, but it doesn't support 64-bit
systems, doesn't have good documentation and is more
I have a delay of 0 on SpeecBackGround, but when I enter DTMF, there's an
almost-exactly five second delay before it returns. Where is this
delay controlled? How can I shorten it?
Is there a way to set the maximum number of digits to look for?
--
This code is really ugly und hard to verify.
Since the computation of the is being done with separate code from the
actual output, the code in that part of the module is indeed ugly. But I
wanted to make the smallest possible change. However, I do suggest that
the full output string be built
Look at SpeechBackground() that comes with Asterisk.
Look here:
http://www.lumenvox.com/help/speechEngineAsterisk/development/dtmf-and-
speech.htm
When you call SpeechBackground() to perform speech recognition, Asterisk
listens for both speech and DTMF entry. As soon as it detects a
Here is a snippet from my lumenvox dialplan (works pretty much the same for
Vestec)
Thanks for the confirmation and sample.
Sorry to be dense, but you're saying that the DTMF comes back in
SPEECH_TEXT(0)? What about SPEECH_SCORE in that case? And what's the
exact difference with Vestec since
I installed the Vestec system and am testing out using it to get strings of
digits (e.g. conference numbers). The sample grammer just allows saying
zero, but almost everybody will read it it oh. But when I try to
add that as an alternative in the grammer (either the word oh or
phonetically as
This one works on my box (Vestec on 1.4.30 on OpenSuse)
Hmm... Not for me.
$Digit = (ONE:1 |
TWO:2 |
THREE:3 |
FOUR:4 |
FIVE:5 |
SIX:6 |
SEVEN:7 |
EIGHT:8 |
NINE:9 |
(OH|ZERO):0);
This is basically the first thing I tried. At least for my voice, this
gets whole lot of spurious 0's.
I have an Asterisk 1.4.2 system online and have built up quite a large
blacklist of tele-spammers that have been calling us. Recently we
swapped one of our DID numbers to a SIP provider that now prefixes all
calls with +1 in front of US numbers (we're in the USA) and
I need
Which speed recognition products will also recognize DTMF? In other words,
I want to say Please speak or dial the conference number. Does Vestec
allow that? LumenVox? Any other way?
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On Mon, May 10, 2010 at 7:19 PM, Richard Kenner ken...@gnat.com wrote:
Which speed recognition products will also recognize DTMF? In other words,
I want to say Please speak or dial the conference number. Does Vestec
allow that? LumenVox? Any other way?
You're on your own for making
I think Asterisk will detect the dtmf for you and the speach recognition will
detect speach.
That's what I was hoping could be done. How do you set up the dialplan
to have both of those functions run simultaneously?
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--
I read the wiki and see mention about needing to set call-limit in
asterisk 1.4 but that has been depreciated in 1.6 so what is the way it
should be done in 1.6?
I set
callcounter=yes
in sip.conf.
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Am I missing something here? I see
if (needvideo) { /* only if video response is appropriate */
add_line(resp, m_video-str);
add_line(resp, a_video-str);
add_line(resp, hold); /* Repeat hold for the video stream */
} else if
I can confirm that the following fixes my problem:
--- chan_sip.c (revision 261450)
+++ chan_sip.c (working copy)
@@ -10357,12 +10357,22 @@
strlen(connection) + strlen(session_time);
if (needaudio)
len += m_audio-used + a_audio-used + strlen(hold);
+
Is there anything special that has to be done to make video calls work?
It doesn't seem to work for me (no video).
What CODECS are supported?
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New to
The Asterisk Development Team has announced the release of Asterisk 1.6.2.7.
What version of Skype for Asterisk works with this release?
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New to
Should be the latest available on the Digium downloads site. It says
version 1.6.2.0 but I've been using Skype for Asterisk on my 1.6.2
branch for quite some time (I just updated it last week).
Hmm. So was I until it abruptly stopped working. It started again when
I went back to an older SVN
Is there an issue with running it with the latest from the 1.6.2 branch?
I did an svn update and make install and now when somebody comes in via
Skype, I get an infinite loop of:
[Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed:
Invalid argument
[Apr 27 09:53:29]
We are running Asterisk 1.6.2.7-rc1 and SfA without problem. What
version are you running?
I'm using the current version from the 1.6.2 SVN branch, which is
called SVN-branch-1.6.2-r258676M. I'm glad to know that 1.6.2.7-rc1 works
because that's closer to what I have than 1.6.2.6.
--
[sip.broadvoice.com]
...
[broadvoice]
exten = 551234,1,Set(CDR(accountcode)=44)
and Asterisk is still giving me this error in the logs (while playing a
number does not exist sound clip):
[Apr 27 18:11:19] NOTICE[12179] chan_sip.c: Call from '551234' to
extension
Is there an explanation other than the one in the application documentation of
exactly what this is for and when you'd want to use it and when you wouldn't?
I find the explanation in the documentation a little confusing.
--
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--
I'm trying to build it and run into all sorts of problems. First,
make testexpr2 doesn't work at top level, nor in the main
subdirectory. If I manually try the commands for it in main/Makefile,
it doesn't have a main and calls ast_log. If use -DSTANDALONE2
instead, those go away, but then:
Why aren't you using check_expr in the utils directory?
Aren't they two different things? I thought check_expr looks at a whole
file for syntax errors while testexpr2 just parses one expression and
returns its value. But if testexpr2 doesn't exist anymore, shouldn't
the documentation be
This begs the question of when the actual violation occurs. In other
words, is this really a usage issue, or does the violation occur
at install time even though the non-GPL component is not usable?
It's hard to see how the violation could occur unless and until the
resulting program were
I'm confused. What does Asterisk do when it gets a 302 with a new number to
forward to? Is there anything I have to do in the dialplan to make this work?
I can't find any clear documentation on this issue.
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You need promiscredir set to yes on sip.conf
And then what do I do in the dialplan? I.e., what context is the
redirect number interpreted in? Google searches on this issue show
inconsistent and contradictory information.
--
After setting promiscredir set to yes * is goign to send the call to
the first desrination on the Contact header.
In case others run into the problem, the fix was not to set promiscredir to
yes in sip.conf, but instead to set FORWARD_CONTEXT in the globals section
of the dialplan.
--
exten = test.skype/example.skype,1, NoOp(nothing)
exten = test.skype/example.skype,n, Hangup()
As you can see, the . (dot) is disappeared and, of course, CID matching
doesn't work as I aspected.
I've try to escape . with something like that \., but nothing.
It seems that asterisk doesn't
exten = _test.,1,Goto(some_context,${FILTER([a-z][0-z],${EXTEN})},1)
I think there's some sort of bug or misfeature here, but I gave up trying
to see exactly what it was.
That filter line probably does not do what you think it does. I would
suggest checking the documentation.
Oops,
I see the following: a stuck process
12651 ?S 0:00 gzip -9 /var/log/asterisk/messages.2
and then:
asterisk*CLI core show channels
Channel Location State Application(Data)
Logger/rotates...@default:1 Down(None)
1. Any chance you're out of disk space?
Nope:
FilesystemSize Used Avail Use% Mounted on
/dev/mapper/VolGroup00-LogVol00
11G 5.2G 4.9G 52% /
/dev/sda1 99M 38M 56M 41% /boot
tmpfs1002M 0 1002M 0% /dev/shm
2. Why not
Is there something strange about using regular expressions in the context
to which incoming Skype calls go?
If I set up accounts, foobar1, foobar2, etc, it doesn't seem to work to
have:
exten = _foobarX,1,...
should it?
--
where in the .call file and format to call cepstral and then the txt
for the message.
Application and Data, respectively.
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asterisk-users mailing
Here's part of the output of running an AGI file:
-- Playing 'degrees' (escape_digits=) (sample_offset 0)
-- Playing 'fahrenheit' (escape_digits=) (sample_offset 0)
-- Playing 'wx/humidity' (escape_digits=) (sample_offset 0)
-- DAHDI/1-1 Playing 'digits/40.ulaw' (language 'en')
I would love to hear some inputs on Aastra and Snom IP phones.
I'm using Aastra 57i phones and like them. They can provisioned easily
(without ANY entries from a local network). The support BLF and I'm also
using the XML capability.
--
The PBX that I'm connecting to Asterisk has a timeout on calls on its PRI
and QSIG lines. But that's smaller than the time it can take some SIP
trunk providers to complete the calls, so I get hangups.
I verified that sending Progress every few seconds will work around the
problem. So I'd like
Is there a version of the Asterisk core sounds in English done by June
Wallack? Some folks here prefer her voice to Allison's, but we'd like
consistency. And is there a version of the Cepstral software with her
voice?
--
_
--
Any idea what can cause this?
asterisk*CLI core show channels
Channel Location State Application(Data)
Logger/rotates...@default:1 Down(None)
1 active channel
0 active calls
20229 calls processed
asterisk*CLI
Is there a way to make a virtual extension busy programmatically?
I want to be able to turn lights on and off on a Polycom phone from a script.
That's what the Custom device type is for.
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That's what the Custom device type is for.
please elaborate I would like to know too
See http://www.voip-info.org/wiki/view/Asterisk+func+device_State
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What's this:
-- Attempting call on DAHDI/g1/9removed for application Wait(5) (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- Channel 0/2, span 1 got hangup, cause 44
-- Forcing restart of channel 0/2 on span 1 since channel reported in use
-- Hungup 'DAHDI/2-1'
How does Asterisk select which of its IP addresses to use to send as the
address to use for RTP connections? I want to be able to use a specific one.
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I'm getting dozens of these at a very high rate:
[Jan 20 09:15:27] NOTICE[16958]: chan_sip.c:12687 check_auth: Correct auth, but
based on stale nonce received from ' sip:1...@gnat.com;tag=as5f1a9480'
[Jan 20 09:15:28] NOTICE[16958]: chan_sip.c:12687 check_auth: Correct auth, but
based on stale
And, I'd be in the camp that would advocate getting your hands dirty and
learn to program without the GUI. You'll learn a lot and then if you'd
want to move to a GUI and something breaks, you'll have some idea on
what and how to fix it.
Knowing now what I do, I find a GUI to
Your comments both come from having taken a short look at FreePBX and
dismissed it without investigating how powerful it can be.
Yes, but the discussion is about COMPLEXITY, not power!
Sure, there are hooks where you can do anything you want, but if you
were to set up identical configurations
I can't seem to find it. Does anybody know where it is?
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It is frustrating to me as we are encouraged to upgrade due to security
issues but if we want to use this particular Digium product we cannot. I
have chosen to upgrade as we have not purchased Fax for Asterisk and as
we are unable to evaluate it I doubt we will. (Not to be snarky but I
don't
Am I correct that if I'm running an -rc or from an SVN release tree that
there's no way I can use any commercial add-ons from Digium, such as
Skype, Cepstral, or G.729?
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Am I correct that if I'm running an -rc or from an SVN release tree
that there's no way I can use any commercial add-ons from Digium, such as
Skype, Cepstral, or G.729?
No, happily not correct. :-)
Digium tries to make their add-ons work with all major releases of
Asterisk. You
You should not try to mix modules for different major versions of
Asterisk. 1.6.0.x modules should only be used with 1.6.0.x, etc.
While John's previous comments were not incorrect, it is unfortunately
quite common that there are API/ABI changes between major releases that
necessitate
I'm using asterisk meetme function like:
exten = 9070,n,MeetMe(|dcM)
and everything works pretty well. But I would like to add a review of
the entered conference number before the user jumps into the conference.
Somthing like:
*:Please enter the conference number followed by the
I made a patch to app_dial.c to make Dial(ext1ext2ext3,tumeout,B)
return busy when just one extension is busy.
Forgive me for the question, but /why/ do you want this behaviour?
Isn't the whole point of dialling multiple extensions so that a call has
a greater chance of being
I have a SIP phone calling an AGI application. It starts out this way:
-- Executing [...@macro-call-agi:2] AGI(SIP/151-b414f0c8,
computer-temp.sh,darwin,) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/computer-temp.sh
Then I get a dozen or so copies of:
[Nov 30
What version of Asterisk are you running? This sounds similar to an
issue with AGI's I saw a while ago, but I can't quite remember
exactly what the issue (or issue number) was.
1.6.2.0-rc2
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On a closely related note, has anyone built a normal (not embedded)
system on SSD?
I've been running Asterisk on a 20GB SSD drive for a while now.
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What mft/model?
Actually, it's 16GB, not 20GB. It's a Transcend TS16GSSD25S-S.
I know that CF cards have a limited number of writes before frying.
If we keep it from using swap am I really only concerned about
voicemail and logs?
That number is quite large, though. I'm taking backups and
And? Noticed any significant performance advantage?
I never ran it any other way, so have no comparison point. I didn't do it
for performance reasons, but reliability.
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Hi,
there are several possibilities do to it
REGISTER Username/Extensions Enumeration
INVITE Username/Extensions Enumeration
OPTION Username/Extensions Enumeration
for more information:
http://www.hackingvoip.com/presentations/sample_chapter3_hacking_voip.pdf
rich...
On Thu, Nov 19, 2009 at
Hi,
asterisk version is 1.4.13
rich...
On Tue, Nov 10, 2009 at 7:01 AM, Tilghman Lesher tles...@digium.com wrote:
On Monday 09 November 2009 15:38:54 Coco Richard wrote:
i'm not sure to understand. Asterisk does support SIP INFO, so why
doesn't Asterisk add the INFO Method in the 200OK
I took a look in chan_sip.c an for 1.4.13 ALLOWED_METHODS doesn't add
INFO. So I will upgrade to 1.6...
thank you for the replies...
rich...
On Tue, Nov 10, 2009 at 9:21 AM, Coco Richard
richard.kingc...@gmail.com wrote:
Hi,
asterisk version is 1.4.13
rich...
On Tue, Nov 10, 2009 at 7
be
included in the list of methods in the Allow header field, when
present.
My SIP provider seems to refuse to send SIP INFO DTMF and releases the
call, because in 200 OK from * there is no INFO Method in the Allow
Header.
Is that correct.
thx
richard
Hi Alex,
i'm not sure to understand. Asterisk does support SIP INFO, so why
doesn't Asterisk add the INFO Method in the 200OK Response?
richard
On Mon, Nov 9, 2009 at 6:38 PM, Alex Balashov abalas...@evaristesys.com wrote:
Yes, it's correct. Asterisk needs to advertise its support
If I have a SIP provider (in this case a PBX using SIP trunks), and
I want to send the calling extension number and name as the from in
the SIP invite, how do I set up my sip.conf entry for that provider? I
find the documentation confusing on this point.
callerid=Some Name In From Header 7065551212
So the first part is the NAME and the second the number, right?
But my question was how to have that be information from the CALLERID
channel variable rather than a fixed value in sip.conf.
___
--
Is this patch correct? The doesn't make logical sense to me. I think
it should be || and making this change fixes the problem I have with SIP
phones in MeetMe conferences. If it's correct, is there someplace more
formal that I should submit it to?
*** app_meetme.c.old2009-10-11
David Backeberg wrote:
From a quick glance at your patch, I would say that it probably tries
to address the audio quality problems I and others were experiencing.
No, it's fixing a much more serious issue. As I sent to this list twice,
when I have a conference between Dahdi ports and SIP
What version are you running?
1.6.2.0-rc2
Does that version support disabling talker optimization?
Yes.
Have you tried disabling talker optimization?
Yes. That's how I found the bug. I got no audio from the SIP phone
into the conference, so I decided I'd try seeing if it did if the SIP
I sent a query about this before, but have some further information and am
hoping somebody has a suggestion as to what to try next to debug this.
I'm using an Asterisk box primarily for MeetMe conferencing. There are
two sources: TDM via two Q.SIG T1's and SIP phones. Conferencing works
fine
Robert McGilvray wrote:
You can do this in the dialplan. Just launch MeetMe with different
options based on the caller,
What's confusing me is that when I look in app_meetme.c, the relevant
options are stored in what are called conference flags and there are
separate user flags. This makes it
We've started to use Asterisk for conferencing and have been getting some
complaints. Our configuration is that some people call in from home, but
we have a physical conference room with a Polycom. When somebody was giving
a presentation in the physical conference room, we were told that the
How do I properly quote things when I want to use the IF function on
something returning a string with blanks (e.g, CALLERID(name))?
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Richard Kenner wrote:
How do I properly quote things when I want to use the IF function on
something returning a string with blanks (e.g, CALLERID(name))?
Use double quotes around your variable
Thanks. That was my second try, but I thought that it didn't work
because I introduced a typo
I'm using QSIG between Asterisk and an NEC SV8300. Whenever I make a call
from the SV8300, I see:
[Oct 4 21:02:49] ERROR[5729]: chan_dahdi.c:12226 dahdi_pri_error: !! Unknown
IE 50 (cs5, len = 3)
I see an IE 50 in the Q.932 specification, so I don't understand why
this error is occuring.
, 0x10, '
Richard home' ]
PROTOCOL 1FI
ASN.1 dump
Context Specific/C [10 0x0A] AA Len:6 06
Context Specific [0 0x00] 80 Len:1 01
00 - ~
Context Specific [2 0x02] 82 Len:1 01
00 - ~
Context Specific/C [1 0x01] A1 Len:24 18
Integer(2 0x02) 02 Len:1 01
00
My system is linked to a legacy PBX via Q-SIG and most of my tests so
far have been from that PBX. I created a number of MeetMe conference rooms
and they work fine when called from the legacy PBX. However, when there's
a MeetMe room with a legacy caller and a SIP phone, the SIP phone can
hear
I'm using QSIG between an NEC SV8300 and Asterisk (after giving up with
CCIS). Things work pretty well with the exception of issues on stations
on the SV8300.
When I call from Asterisk to a SV8300 station and I send my extension
as the caller ID number, it shows up on the SV8300 as OPERATOR.
Hi all,
our asterisk is connected to a sip proxy through a sip trunk. Let's say we
have following dial plan (only an example)
[from_sip_proxy]
exten = 36122512,1,Answer()
exten = 36122512,2,VoiceMailMain()
exten = 3612252,1,Answer()
exten = 3612252,2,MeetMe(313,MI)
exten = 3612252,3,HangUp()
Hi Olle and co
I'm really struggling to convert this into a feature request.
Can anyone help?
Regards,
Richard
--
Richard Brady
T: +44 (0)7771 623 348
E: rnbr...@gmail.com
2009/4/3 Richard Brady rnbr...@gmail.com:
Agreed Olle, it would definitely have to be option driven, not least
-1
On Sat, Jun 13, 2009 at 10:13:11PM -0400, Richard McNeilly wrote:
More Troubleshooting
Ironhide*CLI zap show channels
?? Chan Extension? Context Language?? MOH Interpret
?pseudo??? phones default
? 2??? phones default
am certain that I have the power
connected to the PCI card correctly.
Any suggestions as to what I may be doing wrong here?
Richard
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is not being provided by the
hold initiator.
quote/
For more info see http://www.sipforum.org/sipconnect
Regards,
Richard
On Fri, Apr 3, 2009 at 11:16 AM, Richard Brady rnbr...@gmail.com wrote:
Agreed Olle, it would definitely have to be option driven, not least
for backward compatibility.
When you
Exvito
Did you ever make any progress on this?
Richard
On Mon, Mar 10, 2008 at 2:38 AM, Ex Vito ex.vitor...@gmail.com wrote:
Hi list,
I'm planning and testing a distributed asterisk deployment
throughout several sites; each will be connected to the PSTN
and all of them among
or per device in
channel configurations - or per PBX, also in channel configs.
local hold or remote hold might mean something else, coming to
think of it. But it fitted
in nicely here :-)
/Olle
2 apr 2009 kl. 15.05 skrev Richard Brady:
Furthermore, the following two IETF documents address
address more complex scenarios and solutions,
but they do back me up on the fact that there are good reasons to both
signal hold and provide music.
R.
On Wed, Apr 1, 2009 at 6:16 PM, Richard Brady rnbr...@gmail.com wrote:
Hi Tony
I can see where you guys are coming from on this and have already
behave as I described (Nortel
BCM50, Aastra Intelligate, Mitel 3300 to name a few).
Regards,
Richard
I have to agree with Kevin on this one.
I fail to understand how you have a PBX-A talking to Asterisk talking to
PBX-B and the PBX-A placing the call on hold. Typically you should have
stream straight on.
Any help greatly appreciated.
Richard
On Mon, Mar 30, 2009 at 9:04 PM, Kevin P. Fleming kpflem...@digium.com wrote:
Richard Brady wrote:
If Asterisk is bridging a call between two SIP peers and one peer puts
the other on hold by means of a re-INVITE with SDP containing
, there is a need to get the
scenario above working. How could we go about that?
On Tue, Mar 31, 2009 at 12:45 PM, Kevin P. Fleming kpflem...@digium.com wrote:
Richard Brady wrote:
I have researched the musiconhold / musicclass options in sip.conf as
well as the various documented classes
to change that?
(This is understandable if the peer is a handset but can be a problem
if it is a PBX with its own MOH source.)
Richard
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bilal ghayyad wrote:
And is there a bank accept to give such kind of communication?
The user was able to dial his card number and the amount from his phone (or
IP Phone registered with Asterisk), and Asterisk communicate with the bank or
company credit card provider?
How the user will
Philipp Kempgen wrote:
*snipped
But I guess it wouldn't hurt to add a DEFINED() function to
Asterisk.
if (DEFINED(myvariable)) {
// ...
}
Isn't that what ISNULL is for?
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Philipp Kempgen wrote:
Richard Lyman schrieb:
Philipp Kempgen wrote:
But I guess it wouldn't hurt to add a DEFINED() function to
Asterisk.
if (DEFINED(myvariable)) {
// ...
}
Isn't that what ISNULL is for?
No. ISNULL() works on values
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