When incoming DATA call arrive on ISDN BRI, asterisk (zaphfc) recognise
type of call, but answering anyway (playing IVR messages, ringing
phones, etc...)
How to stop that? I want that only VOICE calls are answered, and
DATA/FAX to be ignored.
(I'm using Asterisk 1.2.1 Brisftuffed 0.3.0-PRE-1f, Z
I can't believe that nobody have this problem. Maybe you just didn't
notice that problem, and your incoming DATA calls are also answered by
IVR.
Somebody?
- Original Message -
From: "Pisac" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Comme
When incoming DATA call arrive on ISDN BRI, asterisk (zaphfc) recognise
that this is DATA call, but answering anyway (playing IVR messages,
etc...)
How to stop that? I want that only VOICE calls are answered, and
DATA/FAX to be ignored.
(I'm using Asterisk 1.2.1 Brisftuffed 0.3.0-PRE-1f, ZapHFC)
When incoming DATA call arrive on ISDN BRI, Asterisk recognise that this
is DATA call, but behaving like it is VOICE call: Answering call,
playing IVR messages...
How to stop that? I want that only VOICE calls are answered by Asterisk
IVR, and DATA/FAX to be ignored.
(I'm using Asterisk 1.2.1 Bri
When incoming DATA calls arrive on ISDN, Asterisk recognise that this is
DATA call, but behaving like it is voice call: Answering call, playing
IVR messages, etc...
How to stop that? I want that only VOICE calls are answered by Asterisk,
and DATA/FAX to be ignored.
(I'm using Asterisk 1.2.1 Brisf
When incoming DATA calls arrive on ISDN, Asterisk recognise that this is
DATA call, but behaving like it is voice call: Answering call, playing
IVR messages, etc...
How to stop that? I want that only VOICE calls are answered by Asterisk,
and DATA/FAX to be ignored.
(I'm using Asterisk 1.2.1 Brisf
I compiled it, and it's WORKING.
Thanks.
But, I would realy realy realy like that somebody explain to
me how is exactly that bug hidden in those two segments?
Where is difference?
Anybody?
1)
if (!IS_DIGITAL(ast->transfercapability))
{set_actual_gain(p->subs[SUB_REAL].zfd, 0, p->rxgain,
Thanks
- Original Message -
From: "Kevin P. Fleming" <[EMAIL PROTECTED]>
> Pisac wrote:
> > make what?
> >
> > If I type make in /asterisksource/channels, then all modules will be
> > compiled, but I tried and I'm getting errors also.
&
- Original Message -
From: "Kevin P. Fleming" <[EMAIL PROTECTED]>
> Pisac wrote:
> > One question, if I change chan_zap.c, what should I type to compile
and
> > install only that module, and not whole asterisk again.
> >
> > I tried
> > gcc chan_
I was never read documentation about that substring functionality. I was
(once upon a time) just thinking about how can I extract first 3 digits
from CALLERIDNUM, and what appears logical to me is that I should try
CALLERIDNUM::3
Guess what, it's worked! That was the first idea that come in to my
One question, if I change chan_zap.c, what should I type to compile and
install only that module, and not whole asterisk again.
I tried
gcc chan_zap.c -o /usr/lib/asterisk/modules/chan_zap2.so
but I'm getting error during compiling.
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You are right, only outgoing calls!
I found lines that you mentioned, but I do not understand where is
difference? In current chan_zap.c I read:
if (!IS_DIGITAL(ast->transfercapability)) {
set_actual_gain(p->subs[SUB_REAL].zfd, 0, p->rxgain, p->txgain, p->law);
} else {
set_actual_gain(p->subs[SUB
Do bristuffed zaptel (zaphfc) supporting rxgain/txgain in zapata.conf?
I'm changing rxgain in zapata.conf, and reloading zaptel, but sound
level on ISDN(HFC) is always the same (loud).
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I'm very surprised that Asterisk (PBX !) do not support AOC.
Setting some variable with AOC informations should be enough.
Storing AOC in CDR would be perfect.
P.
- Original Message -
From: "Armin Schindler" <[EMAIL PROTECTED]>
> On Sun, 15 Jan 2006, Pisa
I'm using bristuffed Asterisk with ISDN/ZAPHFC
I have VERY ugly (outgoing) sound through ISDN/HFC if echocancel=yes in
zapata.conf, but without echocancel I have bad (incoming) echo
Through PSTN/FXO sound is ok with or without echocancel.
I tried other echo cancellers (in zconfig.h) two times:
EC
Do Asterisk support "Advice Of Charge" (AOC) on ISDN lines?
Do any ISDN drivers (bristuff, capi, vISDN, mISDN) support AOC?
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th unexpected line hangup occuring only
when digittimeout=0 and some DTMF digit is pressed during playing some
voice file. IS THIS A BUG?
***
My temporary solution is to set digittimeout=1.
Any comment about this issue?
Cheers.
- Original Message -
From: "Pisac"
r uncomment this:
[options]
;silence_suppression=yes
And see if that helps. You need a timing source for it
to work, which is why it is disabled by default, but the
logging might be a bit chatty in any case.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Beha
I've found something here: http://bugs.digium.com/view.php?id=5374
but I don't understand how this can be connected to my problem :-(
- Original Message -
From: "Pisac" <[EMAIL PROTECTED]>
> I upgraded from 1.0.9 to 1.2.1.
> In 1.0.9 everything worke
Asterisk 1.2.1-BRIstuffed-0.3.0-PRE-1f
- Original Message -
From: "BJ Weschke" <[EMAIL PROTECTED]>
Where did you download this 1.2.1 version of Asterisk from? These
messages are coming from a patch to Asterisk that should not be in any
version of the 1.2 branch.
I upgraded from 1.0.9 to 1.2.1.
In 1.0.9 everything worked perfect.
Now, I call in my IVR, and after navigating in menus when I get dialtone
for dialing extension, Sound is choppy and I get bunch of messagess:
-- Silence suppression is disabled (option_silence_suppression=0
chan->timingfd=30)
--
Yes, you are right, it's working.
Thanks.
- Original Message -
From: "Steve Ringwald" <[EMAIL PROTECTED]>
> Pisac wrote:
> > Sorry, I use correct syntax in dialplan, but here in e-mail I maked
this
> > mistake.
> > In dialplan I'm using
I'm using Asterisk 1.2.1-BRIstuffed-0.3.0-PRE-1f
Your answer was helpfull, it's working now like it used before.
But I'm dissapointed with all this minor & needless & problematic
changes which needlessly spending my time. I will realy double rethink
in the future about upgrading any tuned system
I maked mistake in my previous e-mail, but in my dialplan I didn't make
this mistake. So, my intention in previous e-mail was to write:
${CALLERIDNUM:3} erase first 3 digits
${CALLERIDNUM::3} returns first 3 digits
${CALLERIDNUM:3:3} should erase first 3 digits and return next 3 digits
So,
if
${C
is
placed,
> using your examples:
>
> ${CALLERIDNUM:3} erase first 3 digits
> ${CALLERIDNUM::3} returns first 3 digits
> ${CALLERIDNUM:3:3} should erase first 3 digits and return next 3
digits
>
>
> Pisac wrote on Saturday, 14 January 2006 5:10 AM:
>
> > No,
> &
; <[EMAIL PROTECTED]>
I see this problem too.
Send to us your extensions.conf
there is a diference from extensions in 1.0.X to 1.2.X
ex
exten => _X.,1,Dial(ZAp/g1/${EXTEN}),Ttr
to
exten => _X.,1,Dial,ZAP/g1/${EXTEN},Ttr
try change it.
Rafael Marconi
Em 14/01/2006, às 01:46, Pisa
No,
${CALLERIDNUM}:3 erase first 3 digits
${CALLERIDNUM}::3 returns first 3 digits
${CALLERIDNUM}:3:3 should erase first 3 digits and return next 3 digits
So,
if
${CALLERIDNUM}=0123456789
Then
${CALLERIDNUM}:3 returns 3456789
${CALLERDINUM}::3 returns 012
${CALLERIDNUM}:3:3 returns 345
But this d
I upgraded from 1.0.9 to 1.2.1
My IVR which worked perfectly on 1.0.9, now hangup with no reason (at
least I could not find a cause)
When this hangup happen, I can read:
== Auto fallthrough, channel 'IAX/user-20' status is 'BUSY'
This happening also with ZAP channels
I'm really disappointed with
I'm using point-to-multipoint BRI ISDN, but this simply do not work.
zapCD(number) just have no effect
:-(
Where I can find some documents for bristuff, and zapCD?
- Original Message -
From: "Torsten Krueger" <[EMAIL PROTECTED]>
> Hello,
>
> Giovanni Miano schrieb:
> > call deflect
I upgraded from 1.0.9, to 1.2.1.
I was using this line
exten => s,1,gotoif($[${CALLERIDNUM::3} = 066]?mycity,1:other,1)
it selecting calls if callerid begins with some number pattern (from
some city)
But, it's not working anymore in Asterisk 1.2.1
when I test this with
noop(${CALLERIDNUM::3})
I g
Hey, I tested this today, and it's working!
Thanks!
> I think, the Group-Function in Asterisk 1.2 is what You are looking
for.
> In older versions the Group()-Function was implemented as application
> SetGroup.
> More information can be found in the wiki:
> (http://www.voip-info.org/wiki-Asteris
mailcmd to add a -f? Did you used nail/mail instead of
sendmail, in voicemail.conf? Or maybe some .c source changing?
Thanks
Pisac
> I had a similar problem, but I was able to see the message getting
> rejected to rr.com because they were looking up the hostname pbx
> and rejecting it.
Voice-mail messages aren't sent to e-mail address.
I have two Asterisk servers, first one is upgraded from 1.0.RC2 to
1.0.9, and second one is from 1.0.7 to 1.0.9. Both Asterisk have EXACTLY
same "voicemail.conf" configuration, but second Asterisk don't sending
voice mail messages through e-mail!
Do bristuff/zaphfc support CD (Call Deflection)?
How to deflect call (transfer before answering) with bristuff?
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I have ISDN BRI line (Point-to-multipoint), with HFC card and
bristuff/zaphfc driver.
I also have TA attached to NT, with analog phones and modem on it.
When I'm dialling through Asterisk/bristuff, and in the same time TA
have some conversation (or maybe modem link) on channel 1, I can hear
that
I found solution:
Just set "overlapdial=yes" in zapata.conf, and then in extensions.conf
"dial(zap/1/)" or if you using groups "dial(zap/g1/)", and you will get
dialtone from local exchange (telekom).
Cheers.
> It's not working for me. If I make interdigit pause more than 1 sec, I
> get hangup
It's not working for me. If I make interdigit pause more than 1 sec, I
get hangup (busy) if number is not complete.
> I don't know if it's possible, but I use a workaround to simulate the
> external dialtone:
>
> I use '0' to access external lines
>
> exten -> _0,1,ChanIsAvail(Zap/g1)
> exten ->
How can I get external (telecom local exchange) dialtone on HFC ISDN BRI
with bristuff/zaphfc driver?
with capi, voip-info say that it should be something like:
Dial(CAPI/MSN:b)
But with zaphfc, if I try: Dial(ZAP/1/), I just get NOANSWER.
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How can I limit incoming concurent calls per MSN on BRI ISDN?
I'm using bristuff/zaphfc with imediate=no in zapata.conf, and I have 4
MSNumbers. I want that incoming caller (to my MSNumber 1234) get busy if
that number is already in use through another B channel, so that other B
channel is availab
> > I'm reading voip-info... and it's only confusing me:
> >
> > zaphfc, zapbri driver package, bristuff...
> >
> > So, what to download and install? If I install bristuff from
> > junghanns.net, should I also install something else (patch)?
> > What is (and where is) that zapbri driver package?
>
ing kernel 2.4.31
- Original Message -
From:
Giovanni
Miano
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: čet 29. dec 2005 22:31
Subject: Re: [Asterisk-Users] Easiest way
to use HFC-S?
Use Bristuff
2005/12/29, Pisac <[EMAIL PROTEC
What is the easiest way to install and use HFC-S card on Asterisk?
As less kernel compiling & driver installations as possible.
Is it mISDN, or chan_capi, or vISDN, or zaphfc, or?
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Not in CLI, Invoked in extensions.conf:
exten => s,1,system(/usr/bin/logscript) ;and how to pass some parameters
here?
if I do somenhing like:
exten => s,1,system(/usr/bin/logscript,${CALLERID},pstn)
then I get error.
- Original Message -
From: "Pisac" <[EMAIL PROTEC
How to pass some parameters to shell script, invoked in CLI through
application system(...)?
I want to do some logging of incoming CID-s to file. Is there some other
method to do this?
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I have very loud sound through IAX2 channel, very saturated in some
moments.How to find where is problem. I think problem is at provider
side, but how to be doubtless?
Is there any method to measure and change sound level on IAX channel (like
on Zap channel)?
__
exten => 3999,2,VoicemailMain(s${CALLERIDNUM})
if you extension is 104, then it will be converted inside asterisk to:
exten => 3999,2,VoicemailMain(s104)
and that will give to you access to mailbox 104 without password prompt (s=skip) and you can retreive messages.
${CALLERIDNUM} is extension
I don't have CCM4..., but if somebody know how I to get one...
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Yes, but I'm asking for CM3.3 connected via H323.
I can set everything, including voice mail button, automatic forwarding to voice-mail... but only I didn't find a way to turn-on voice-mail lamp on Cisco phone (MWI)
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