.
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this by setting the following in voicemail.conf
; How many seconds of silence before we end the recording
maxsilence=5
; Silence threshold (what we consider silence, the lower, the more sensitive)
silencethreshold=512
HTH
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ringing from the PSTN through a zap card, the rings are
coming from the phone company and are just sound. * doesn't decode that
and act on it yet.)
Maxim Vexler wrote:
On 7/16/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Jul 16, 2006, at 11:36 AM, Maxim Vexler wrote:
Hello list
I'm trying
On 7/19/06, Maxim Vexler [EMAIL PROTECTED] wrote:
On 7/18/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
I think when a PSTN line says 'Ring' it's simply for aesthetics... The
line is 'answered' the instant * connects to it for two-way audio...
(well not that instant but somewhere
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Actually it's quite rational.
Check the paragraph on ECM at http://www.voip-info.org/wiki-Asterisk+fax
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call IVR
after the specified(where?) number of seconds.
Thank you.
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On 7/16/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Jul 16, 2006, at 11:36 AM, Maxim Vexler wrote:
Hello list
I'm trying to setup asterisk as an answering machine.
How can I set asterisk to Answer() incoming call ONLY after specified
count of ring cycles ?
In the current situation I
for helping info.
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: return non zero value) and you used
the operator then the second command won't run. where as ';' simply
means run them all, one after the other.
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tested ?
If so then this is a great tip!
thank you!
Also, http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
could use this tip in a form of a comment / page edit.
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A bit of information would help
Please post the output of :
ls -lAr /dev/zap/
and
ps -ef | grep asterisk
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NVFaxDetect does just that ;)
Any why, you might find it more useful to actually receive the fax :
http://www.voip-info.org/wiki/view/NVFaxDetect
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might want to check pattern matching :
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns
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-users
No it is not.
Please attach the relevant part of the log from /var/log/asterisk/full.
Also, as a reference, you might wish to check FreePBX project can
generate IVR dial logic using a simple web management interface.
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otherwise he gets user to
record a message in our user's Foo voicemail box.
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problem I'm having now is that I must run the script as user
asterisk, otherwise I get the same asterisk error as before.
Is there any way to fix this ?
Thank you.
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: Permission denied,
deleting
May 24 08:57:27 WARNING[10618]: pbx_spool.c:389 scan_thread: Failed to
scan service '/var/spool/asterisk/outgoing/max.call'
What am I doing wrong ?
Thank you.
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this for the 2.6 setup ?
Can it be done from the AMP web managment portal ?
Our setup uses the zapata.conf file for phone configurations, and the
call's are coming from PSTN TDM400 FXO card.
Thank you.
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Can it be done from the user's phone (Sipura 841) ?
Can I as the wwwadmin user set it ?
Thank you.
My knowledge in dialing rules is rather limited (i.e. null...)
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