Hi,
I want to change my voicemail message based on the time of day. I would
like a message that says Sorry the office is now closed. after a
certain time, and says Sorry I am unavailable / Busy / etc before.
I have come up with two ways of doing it:
1. A cron job to replace the files
Hi,
I want to change my voicemail message based on the time of day. I would
like a message that says Sorry the office is now closed. after a
certain time, and says Sorry I am unavailable / Busy / etc before.
I have come up with two ways of doing it:
1. A cron job to replace the files
/index.php?page=Asterisk+tips+openhours
Wildheart wrote:
Hi,
I want to change my voicemail message based on the time of day. I
would
like a message that says Sorry the office is now closed. after a
certain time, and says Sorry I am unavailable / Busy / etc before.
I have come up with two
Hi,
I have a dialplan that works like this:
; Arg 1 is the phone, Arg2 is the timeout (optional), Arg 3 is the
voicemailbox(optional)
exten = 20,1,Macro(dialexten,SIP/1234,15,1234)
exten = 21,1,Macro(dialexten,SIP/1235,15)
; Arg 1 is phones, Arg 2 is timeout, Arg 3 is voicemail
exten =
Hi,
The PSTN connection is via a zaptel card, rather than a sip peer.
With thanks,
Tim
On 2006-10-24 06:44:01 -0700, Wildheart
[EMAIL PROTECTED] said:
Hi,
Does anyone know a what to use a different codec for calls which a
re
handset to handset (eg, G711) then when we have
Hi Marty,
By the outside world, I mean the PSTN connection. I am still interested
in how you would set this up. Can you paste in a sample config?
With thanks,
Tim
On 2006-10-24 06:44:01 -0700, Wildheart
[EMAIL PROTECTED] said:
Hi,
Does anyone know a what to use a different
Hi,
Does anyone know a what to use a different codec for calls which are
handset to handset (eg, G711) then when we have calls to the out side
world (via an asterisk server) to use a different codec(eg, G729)?
The idea is to reduce the bandwidth to the server for the majority of
calls,
be causing it?
With thanks,
Wildheart
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Hi,
Can someone tell me what are the valid parameters for the option
echocancelwhenbridged? Is it just yes or no, or does it support 128 as
well? Also is thier any differnce with using
echocancelwhenbridged=128 as opposed to echocancelwhenbridged=yes
(assuming that 128 is a valid option).
Hi Doug,
Shouldn't you really be using ${CALLERID(number)}?
Also, if the channel you are using to get the caller ID from is analog (FXO
or FXS), I believe you may have to answer the channel, then wait 1 sec to
get the correct caller id info.
Tim
- Original Message -
From: Douglas
D'oh... Might have just answered the wrong question here...
Also, if the channel you are using to get the caller ID from is analog
(FXO
or FXS), I believe you may have to answer the channel, then wait 1 sec to
get the correct caller id info.
Tim
Hi,
I am trying to join 2 asterisk servers together using a sip channel.
This is so, if a user joins a conference on box A and another user
joins a conference on box B, providing they are in the same conference
room, the two conferences are joined via the sip channel. We only want
to join the
Hi,
Have a look at this ticket:
http://bugs.digium.com/view.php?id=6874
It contains the patch to add dbdel to your implimetation, but the
command is not being added to the core of asterisk.
Tim
Hi list,
is there a possibility to delete a key from the astdb through the
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