I have a DID with budgetphone.nl, which has worked fine for quite some
time. For the last few weeks, most (but not all of the time), the incoming
call does not go where it is supposed to, but instead the following message
show on the console:
[Aug 30 19:47:28] NOTICE[5161]: chan_sip.c:13865
that no introductions are to be saved in the priv-callerintros
directory.
N- This option is a modifier for the screen/privacy mode. It
specifies
that if callerID is present, do not screen the call.
On Sun, May 11, 2008 at 12:24 PM, Robert DeVries [EMAIL PROTECTED]
wrote
Are you certain Asterisk is not sending the emails, rather than them not
being received? i have had problems in the past with spam filters
rejecting the emails.
On Tue, May 13, 2008 at 4:48 PM, Roberto Milani
[EMAIL PROTECTED] wrote:
Hello list users
I have a very nice installation of
GrandCentral has a feature where when you call the GrandCentral number it
can ring multiple phones. However, it's not the first phone to answer that
gets connected, but the first phone to answer AND play a touch-tone after
hearing a recording. The advantage of this is that if one of the called
I am having some hardware problems with the Linux machine where I have
Asterisk installed and want to use a different machine.
Assuming I install Asterisk on machine number 2, is it possible to just move
files over from the old machine to the new machine and the new machine will
behave like the
I have a friend traveling overseas. I want to allow him to call a number
which will give him a busy signal (so no charge), but will then send me an
email that he has called.
I know how to use a call file to trigger a call (I created a callback system
for myself when I traveled overseas a few
When using a SIP phone with Asterisk, hitting the # key (pound or hash
depending on where in the world you happen to be) tells Asterisk that there
are no more digits coming, and to put the call through immediately based on
the digits already entered. This is the same functionality as the PSTN
Although probably not what you want to hear, I don't think there is any way
that Asterisk will detect answering supervision on an analog POTS line (I
believe that there used to be an option with some Telcos to get a polarity
reversal upon answering, but I don't think that is possible any more.)
/callback.info
/var/spool/asterisk/outgoing)
and it worked perfectly. Problem solved. Thanks.
On 2/7/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: *Robert DeVries [EMAIL PROTECTED]
I am trying to get called back with a DISA dial tone when I call a trigger
number. I got it to work almost the way I want
I am trying to get called back with a DISA dial tone when I call a trigger
number. I got it to work almost the way I want, this is the callback
context:
[callback]
exten= 501,1,Congestion()
exten= 501,2,Hangup()
exten =h,1,System(cp /etc/asterisk/callback.info
/var/spool/asterisk/outgoing)
I want to essentially transplant my existing Asterisk server to a new
machine, and take the old sever out of service.
Assuming I install Asterisk on the new machine, does anyone know what files
I would have to copy over? What comes to mind are the *.conf files in
/etc/asterisk, as well as the
I have found that if you don't have the minimum balance required for the
voipjet premium server, you get the circuits busy message, you might
want to check your balance.
On 1/30/07, Alejandro Lengua [EMAIL PROTECTED] wrote:
Hello, we have this problem with Trixbox 1.23
I have created an
Anyone know of any?
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They are a bit on the expensive side - I'm looking for something along the usual 2-3 cents per minute.On 8/31/06, www.IPKall.com
[EMAIL PROTECTED] wrote:
www.Kall8.com
IPKall
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of Robert DeVries
I would expect that there might be an additional charge for Caller ID Name Lookup - but not 4 cents a minute or more (in fact, I would assume that the incremental cost is the same regardless of the length of the call.)
On 8/31/06, Jeremy McNamara [EMAIL PROTECTED] wrote:
Robert DeVries wrote
I would like to have callers that call different DID numbers receive different outgoing messages (based on the number called), but have all of the incoming messages in one box. Any way to do this that comes to mind?
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The serveremail setting in voicemail.conf is supposed to change the return address of the voicemail notifications. I have found that it indeed does this successfully for the first email entered (the one intended for sending to a conventional email address), but does not do so for the second email
. However, I can't find any variable on the wiki that
represents ANI in order to test this theory.
Any ideas out there?
On 2/9/06, Robert DeVries [EMAIL PROTECTED] wrote:
I recently upgraded to 1.2.x (not quite sure which version, whatever it is I downloaded it on 1/30/06.)
At the time
I recently upgraded to 1.2.x (not quite sure which version, whatever it is I downloaded it on 1/30/06.)
At the time of the upgrade, I had a Nufone toll-free number going into
my IVR. Extension 2000 rang the Cisco phone on my desk, and the
caller id came through just fine.
As soon as the
I recenlty upgraded to 1.2.x (not quite sure which version, whatever it is I downloaded it on 1/30/06.)
At the time of the upgrade, I had a Nufone toll-free number going into
my IVR. Extension 2000 rang the Cisco phone on my desk, and the
caller id came through just fine.
As soon as the upgrade
I have a number of situations where in the past I would get a native
bridge, IAX to IAX - e.g., call coming in on an IAX VOIP line being
forwarded to PSTN via another IAX termination provider, or calls made
from my IAXY to a PSTN number via my Asterisk box to a VOIP provider.
I find the native
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