[Asterisk-Users] DATA CALLS annoying my system
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, ZapHFC) Log: -- Accepting data call from '' to '3001' on channel 0/2, span 1 -- Executing Answer(Zap/2-1, ) in new stack -- Executing BackGround(Zap/2-1, ivr_intro) in new stack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why incoming DATA CALLS are answered as VOICE by asterisk IVR?
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-Commercial Discussion asterisk-users@lists.digium.com Sent: uto 28. feb 2006 17:26 Subject: [Asterisk-Users] why incoming DATA CALLS are answered as VOICE byasterisk IVR? 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) Log: -- Accepting data call from '' to '3001' on channel 0/2, span 1 -- Executing Answer(Zap/2-1, ) in new stack -- Executing BackGround(Zap/2-1, ivr_intro) in new stack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] why incoming DATA CALLS are answered as VOICE by asterisk IVR?
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) Log: -- Accepting data call from '' to '3001' on channel 0/2, span 1 -- Executing Answer(Zap/2-1, ) in new stack -- Executing BackGround(Zap/2-1, ivr_intro) in new stack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DATA calls answered by IVR, but I don't want that
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 Brisftuffed 0.3.0-PRE-1f, ZapHFC) -- Accepting data call from 'XX' to '3001' on channel 0/2, span 1 -- Executing Answer(Zap/2-1, ) in new stack -- Executing BackGround(Zap/2-1, ivr_intro) in new stack ... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming ISDN DATA calls answered by asterisk IVR! - How to stop that?
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 Brisftuffed 0.3.0-PRE-1f with ZapHFC ISDN BRI lines) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming ISDN DATA calls answered by asterisk IVR! - How to stop that?
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 Brisftuffed 0.3.0-PRE-1f with ZapHFC ISDN BRI lines) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RX/TXgain on bristuff/zaptel ?
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_REAL].zfd, 0, 0, 0, p-law);
}
And your suggestion is:
if (IS_DIGITAL(ast-transfercapability)) {
set_actual_gain(p-subs[SUB_REAL].zfd, 0, 0, 0, p-law);
} else {
set_actual_gain(p-subs[SUB_REAL].zfd, 0, p-rxgain, p-txgain, p-law);
}
Which is the same thing but invertedly written. I'm not a programmer, so
I may be wrong (maybe IS_DIGITAL could be NULL), but I would like to
understand difference in those two segments.
Regards.
P.
- Original Message -
From: Koopmann, Jan-Peter [EMAIL PROTECTED]
Please recheck: This happens on outgoing calls only, correct? On inbound
calls rxgain/txgain work? There is a bug in current bristuff: Somwhere
around line 1928 you should find a few lines regarding gain. Change them
to look like this:
if (IS_DIGITAL(ast-transfercapability)) {
set_actual_gain(p-subs[SUB_REAL].zfd, 0, 0, 0, p-law);
} else {
set_actual_gain(p-subs[SUB_REAL].zfd, 0, p-rxgain,
p-txgain, p-law);
}
Recompile and things will start working on outgoing calls as well.
Kind regards,
JP
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[Asterisk-Users] How to compile and install just one module?
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. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CALLERIDNUM::3 do not working on 1.2.1
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 mind, and it really worked, and (of course) I believed that it is the way it should work. By the way, I understand your point of view, but, when I'm making a new dialplan (IVR system) I don't think about what is documented and what isn't. I just wrote dialplan and try it, and if it working like it should, then I switch my mind to other problems. By the way, autofallthrough is also needless problematic change :-). You should set it as default =NO, and only if in extensions.conf exist autofallthrough=yes it should work. I had problems with this feature, that appears to me as a bug. I still think that is a bug: unexpected disconnecting IVR if I press any digit during playing announce (not after announce) and only if digittimeout=0. H? - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] It was _never_ documented that you could skip a numeric parameter for the substring functionality and expect it to work properly. If it did, you got lucky. When it stopped, you got unlucky. This was not a 'needless problematic' change, it was not a change at all, for people who followed the documentation. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to compile and install just one module?
make what? If I type make in /asterisksource/channels, then all modules will be compiled, but I tried and I'm getting errors also. If I type make chan_zap.o, or chan_zap.c, I'm also getting errors. I even tried gcc -B ../ -o /usr/lib/asterisk/modules/chan_zap2.so chan_zap.c - 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_zap.c -o /usr/lib/asterisk/modules/chan_zap2.so but I'm getting error during compiling. What is wrong with typing 'make'? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to compile and install just one module?
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. If I type make chan_zap.o, or chan_zap.c, I'm also getting errors. You are making this much harder than it needs to be. Just type 'make' in the top-level Asterisk source directory. Everything will work as expected. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Advice Of Charge (AOC) ?
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, Pisac wrote: Do Asterisk support Advice Of Charge (AOC) on ISDN lines? Do any ISDN drivers (bristuff, capi, vISDN, mISDN) support AOC? It is not implemented in chan_capi yet, but this is very easy. The question is what should be done with the AOC information? Just set some variable? As far as I know Asterisk has no API/structure for that. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RX/TXgain on bristuff/zaptel ?
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). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CALLERIDNUM::3 do not working on 1.2.1
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 do not work anymore in 1.2.1, and if I do not found solution
for this I will downgrade to 1.0.9
- Original Message -
From: Dinesh Nair [EMAIL PROTECTED]
i believe the syntax is ${CALLERIDNUM:3} and not as you're using it
with
double colons. also, the present accepted method is to use the
CALLERID()
function rather than the variable which may be deprecated in future
releases.
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
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Re: [Asterisk-Users] auto fallthrough hangup on 1.2.1
It's IVR system with complicated extensions.conf and ivr.conf (included
in extensions.conf), and I don't know which part of it to send here,
because IVR hangup in different places in different times (so it's seems
to me independent from what is written in extensions/ivr.conf, but I
belive it could not be independent, so I must explore this problem
more).
Dial is not a problem, because IVR hangup even if I do not use Dial at
all !
Hangup occur only when I press (any) DTMF digit.
What's happening?
:-(
- Original Message -
From: Rafael Marconi [EMAIL PROTECTED]
I see this problem too.
Send to us your extensions.conf
there is a diference from extensions in 1.0.X to 1.2.X
ex
exten = _X.,1,Dial(ZAp/g1/${EXTEN}),Ttr
to
exten = _X.,1,Dial,ZAP/g1/${EXTEN},Ttr
try change it.
Rafael Marconi
Em 14/01/2006, às 01:46, Pisac escreveu:
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 1.2.1, what is benefit from upgrade if I
must spend couple days to get my system to work as it worked
previously
before upgrade (I think it should be named troublegrade).
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Re: [Asterisk-Users] CALLERIDNUM::3 do not working on 1.2.1
Sorry, I use correct syntax in dialplan, but here in e-mail I maked this
mistake.
In dialplan I'm using ${CALLERIDNUM::3}
- Original Message -
From: Trevor G. Hammonds [EMAIL PROTECTED]
You are using incorrect syntax. Notice where the close bracket 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,
${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 do not work anymore in 1.2.1, and if I do not found
solution
for this I will downgrade to 1.0.9
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Re: [Asterisk-Users] CALLERIDNUM::3 do not working on 1.2.1
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
${CALLERIDNUM}=0123456789
Then
${CALLERIDNUM:3} returns 3456789
${CALLERDINUM::3} returns 012
${CALLERIDNUM:3:3} returns 345
But this do not work anymore in 1.2.1, and if I do not found solution
for this I will downgrade to 1.0.9
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Re: [Asterisk-Users] CALLERIDNUM::3 do not working on 1.2.1
I'm using Asterisk 1.2.1-BRIstuffed-0.3.0-PRE-1f
Your answer was helpfull, it's working now like it used before.
But I'm dissapointed with all this minor needless problematic
changes which needlessly spending my time. I will realy double rethink
in the future about upgrading any tuned system to new Asterisk release.
Thanks for your help.
- Original Message -
From: trixter aka Bret McDanel [EMAIL PROTECTED]
using SVN-branch-1.2-r7847M+trxtel (no modifications to this part per
the +trxtel stuff) I get
-- Executing NoOp(SIP/31-f1eb, 19164048307) in new stack
-- Executing NoOp(SIP/31-f1eb, 64048307) in new stack
-- Executing NoOp(SIP/31-f1eb, 19164048307) in new stack
-- Executing NoOp(SIP/31-f1eb, 640) in new stack
-- Executing NoOp(SIP/31-4f86, 191) in new stack
from
exten = 4,1,noop(${CALLERIDNUM})
exten = 4,2,noop(${CALLERIDNUM:3})
exten = 4,3,noop(${CALLERIDNUM::3})
exten = 4,4,noop(${CALLERIDNUM:3:3})
exten = 4,5,noop(${CALLERIDNUM:0:3})
The only one that didnt work as you described was the '::3' and once I
implicitly stated '0' as the starting offset it works as you
described.
What version of 1.2 are you using?
--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605 Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
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Re: [Asterisk-Users] CALLERIDNUM::3 do not working on 1.2.1
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 ${CALLERIDNUM::3}
Just for grins, have you tried
${CALLERIDNUM:0:3}
I have always found it better to explicitly specify what to do, rather
than relying on a function's assumptions
Steve
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[Asterisk-Users] 1.2.1 Silence suppression is disabled what the hell?
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) -- Silence suppression is disabled (option_silence_suppression=0 chan-timingfd=30) -- Silence suppression is disabled (option_silence_suppression=0 chan-timingfd=30) -- Silence suppression is disabled (option_silence_suppression=0 chan-timingfd=30) -- Silence suppression is disabled (option_silence_suppression=0 chan-timingfd=30) ... What is this? ggghh :-(( ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.2.1 Silence suppression is disabled what thehell?
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. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.2.1 Silence suppression is disabled what thehell?
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 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) -- Silence suppression is disabled (option_silence_suppression=0 chan-timingfd=30) -- Silence suppression is disabled (option_silence_suppression=0 chan-timingfd=30) -- Silence suppression is disabled (option_silence_suppression=0 chan-timingfd=30) -- Silence suppression is disabled (option_silence_suppression=0 chan-timingfd=30) ... What is this? ggghh :-(( ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.2.1 Silence suppression is disabledwhatthehell?
It's helped. Thanks! - Original Message - From: Dan Austin [EMAIL PROTECTED] I looks like someone decided to bundle a patch that hasn't been merged yet. Good for testing, not so good for initial impressions. In /etc/asterisk/asterisk.conf add or 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 Behalf Of Pisac Sent: Saturday, January 14, 2006 9:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 1.2.1 Silence suppression is disabled whatthehell? 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. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] auto fallthrough hangup on 1.2.1
I isolated problem, but I cannot find a cause. I think this is a bug! So, there is very very simplified dialplan which working in 1.0.9 but in 1.2.1 have that unexpected hangup: ;- exten = s,1,answer exten = s,2,digittimeout(0) exten = s,3,responsetimeout(15) exten = s,4,background(ivr-announce) exten = 1,1,goto(option1,s,1) exten = 2,1,goto(option2,s,1) exten = t,1,hangup exten = i,1,goto(s,1) ;- When I press any digit DURING PLAYING ivr-announce (not after announce is finished), my line hangup: == Auto fallthrough, channel 'IAX2/someusername-2' status is 'UNKNOWN' -- Hungup 'IAX2/someusername-2' Where 'IAX2/...' is channel through I connected to IVR. If I connect through ZAP (ISDN/PSTN) then there is written 'ZAP/...', and if I get to this IVR context after some dialing of busy number, then ...status is 'BUSY' instead 'UNKNOWN'. If i change digit timeout to 1 sec: exten = s,2,digittimeout(1) then everything working as it should !!! *** So, conclusion is that problem with 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 [EMAIL PROTECTED] 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 1.2.1, what is benefit from upgrade if I must spend couple days to get my system to work as it worked previously before upgrade (I think it should be named troublegrade). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Advice Of Charge (AOC) ?
Do Asterisk support Advice Of Charge (AOC) on ISDN lines? Do any ISDN drivers (bristuff, capi, vISDN, mISDN) support AOC? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ugly echo cancel, with Bristuff/Zaphfc
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: ECHO_CAN_KB1 (this was default) ECHO_CAN_MARK2 ECHO_CAN_MG2 after any change I compiled (make clean all make install) all drivers which are bristuffed, without recompiling Asterisk (should I?), but... ... I always have EXACTLY same (ugly) sound and same (appearing of) echo cancelling. Seems to me that echo canceller do not changing no matter what I choose in zconfig.h. This must be some bristuff echo cancel patch to zaptel source? Before I bristuffed Asterisk, I was playing with zconfig.h and I remember that difference between echo canceller were very obvious. Anybody have some knowledge about bristuffed echo cancelling? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CALLERIDNUM::3 do not working on 1.2.1
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 get full callerid, not just first 3 numbers like it use to be on 1.0.9
Why?
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Re: [Asterisk-Users] CD (call deflection) on Bristuff/zaphfc?
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 deflection does not work with bristuff this is no longer true - at least not when using a recent bristuff version and a point-to-multipoint trunk. exten = 37,1,Wait(0.5) exten = 37,2,ZapCD(destination-number) exten = 37,3,Progress() exten = 37,4,4,Hangup Does just what it should. Unfortunately this does not work when using a point-to-point connection. In this case you would the facility 'reroute' and this is not implemented in bristuff. BTW, the Sirrix channels can also do both. Regards Torsten ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] auto fallthrough hangup on 1.2.1
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 1.2.1, what is benefit from upgrade if I must spend couple days to get my system to work as it worked previously before upgrade (I think it should be named troublegrade). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limit concurent calls per MSN on BRI(bristuff/zaphfc)?
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-Asterisk+cmd+SetGroup). I am using something like the code under Example 2 Revised on that page to limit the calls to a sip phone. HTH, Karsten ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice mail messages aren't sent to e-mail
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! I'm using almost default voicemail.conf with just one mailbox addedd: 1234 = 1234,MyName,[EMAIL PROTECTED] Why second Asterisk don't sending e-mails? I tested nail program, and I can send any mail without problems. How Asterisk sending mail, through some other program (nail, mail) or by itself? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice mail messages aren't sent to e-mail
Yes, I found that this is problem with my server. Second server is connected through second provider, and first server and my domain is hosted at fist provider. My (first) provider has some stupid logic that reject e-mails from mailservers which don't have public hostname but private (my second server has server.local), but accepting all e-mails from it's IP address space (first server). So, my temporary solution was that I set up fake (but existing) hostname for second server (gmail.com), and now my (first) provider accepting e-mails. Very stupid. How you changed 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. I changed the mailcmd to add a -f realhostname.com and it started working. - James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] local exchange dialtone on ISDN/bristuff?
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 - _0,2,playtones(dial)
exten - _0,3,goto(external_tone|et)
...extensions if some dialed without waiting for dialtone
[external_tone]
exten = et,1,DigitTimeout(1)
exten = et,2,Playtones(dial)
exten = et,3,WaitExten(8)
exten = _X,1,DIAL(ZAP/g1/${EXTEN})
exten = _X.,1,DIAL(ZAP/g1/${EXTEN})
exten = _X,102,PLAYTONES(busy)
exten = _X.,102,PLAYTONES(busy)
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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Re: [Asterisk-Users] local exchange dialtone on ISDN/bristuff?
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 (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 - _0,2,playtones(dial)
exten - _0,3,goto(external_tone|et)
...extensions if some dialed without waiting for dialtone
[external_tone]
exten = et,1,DigitTimeout(1)
exten = et,2,Playtones(dial)
exten = et,3,WaitExten(8)
exten = _X,1,DIAL(ZAP/g1/${EXTEN})
exten = _X.,1,DIAL(ZAP/g1/${EXTEN})
exten = _X,102,PLAYTONES(busy)
exten = _X.,102,PLAYTONES(busy)
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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[Asterisk-Users] bristuff/zaphfc disturbing other ISDN phones
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 conversation (or modem) very short period (0.2sec), and I also disturbing that conversation (modem). On CLI is shown: -- Called g1/454325454 -- Moving call from channel 1 to channel 2 But this short moving call from busy channel 1 to free channel 2 is enough to break my modem link which operating on channel 1. Interesting, if I call exactly through channel 1 dial(ZAP/1/...) I'm also getting Moving call from channel 1 to channel 2. It's no help to use chanisavail(ZAP/1) because same thing happening: channel is marked as available, but then disturbing existing conversation on channel 1 and then moving from 1 to 2. If channel 1 is occupied with Asterisk (bristuff) conversation, then this problem don't exist (no disturbing). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CD (call deflection) on Bristuff/zaphfc?
Do bristuff/zaphfc support CD (Call Deflection)? How to deflect call (transfer before answering) with bristuff? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] local exchange dialtone on ISDN/bristuff?
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. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Limit concurent calls per MSN on BRI (bristuff/zaphfc)?
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 available for incoming calls to other MSNumbers. I don't want that 2 incoming calls to MSN 1234 invade both B channels and congest my ISDN. Any solution? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Easiest way to use HFC-S?
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? Go to the junghanns.net page, get the latest bristuff. Unpack. You will find among other things a readme file explaining what to do. -- Best regards Peer Oliver Schmidt Do I need to use this download.sh script in bristuff? I already have working Asterisk (same version), so why to download and install again? Can I only patch sources and then compile? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Easiest way to use HFC-S?
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? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Easiest way to use HFC-S?
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? Im using kernel 2.4.31 - Original Message - From: Giovanni Miano To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: et 29. dec 2005 22:31 Subject: Re: [Asterisk-Users] Easiest way to use HFC-S? Use Bristuff 2005/12/29, Pisac [EMAIL PROTECTED]: 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?___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] System(...) but how to pass parameters?
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? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] System(...) but how to pass parameters?
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,${CALLERID},pstn)
then I get error.
- Original Message -
From: Pisac [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: ned 25. dec 2005 1:08
Subject: [Asterisk-Users] System(...) but how to pass parameters?
How to pass some parameters to shell script, invoked in CLI through
application system(...)?
I want to do some logging of incoming CID-s to file. Is there some
other
method to do this?
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[Asterisk-Users] Sound too loud (saturated). How to change?
I have very loud sound through IAX2 channel,very saturated in some moments.How to find where is problem. I think problem is at provider side, but how to be doubtless? Is there any method to measure and change sound level on IAX channel (like on Zap channel)? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to retrieve voicemail from an IP phone?
exten = 3999,2,VoicemailMain(s${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 passwordprompt (s=skip) and you can retreive messages.
${CALLERIDNUM} is extension (caller id)number of caller, and caller gets his own mailbox.
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Re: [Asterisk-Users] Cisco Callmanager Asterisk for Voicemail revisited
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) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Callmanager Asterisk for Voicemail revisited
I don't have CCM4..., but if somebody know how I to get one... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
