Hello List,
I want to play an announcement for attended transfer calls. For example, A
calls B, B answers the call and transfers (attended) to C - once transfer
is complete B should hear an announcement saying you call has been
transferred. Is there any configuration in asterisk to implement
hello everyone
i have a simple question: i have an asterisk which is a h323 gateway
and has a h323 connection to a cisco gatekeeper and a sip connection
to a pbx.
my question is: how can i send all calls to gatekeeper?
i searched a lot and found that i should set gatekeeper=192.168.0.X
(ip
While B is talking to C, A is enjoying MOH. You could install a
musicclass that starts with Your are being
Playing an announcement like Your call has been... to A after C has
accepted the call is probably not a good idea, because C has to wait
until the the announcement has finished. In
B.H.
Hello!
We have several Asterik boxes that are connected to PSTN using PRI cards
and they are interconnected using IAX2 trunks so that incoming calls are
delivered from PSTN to the servers they belong to.
In past we were using asterisk 1.4 on the server that is receiving IAX
connections and
WARNING[] chan_iax2.c: Too much delay in IAX2 calltoken timestamp from
address X.X.X.X
I don't know if this will help, but I have:
requirecalltoken=no
In my iax.conf
Doug
--
_
-- Bandwidth and Colocation Provided
B.H.
On Tue, Jun 11, 2013 at 3:45 PM, Doug Lytle supp...@drdos.info wrote:
WARNING[] chan_iax2.c: Too much delay in IAX2 calltoken timestamp
from address X.X.X.X
I don't know if this will help, but I have:
requirecalltoken=no
In my iax.conf
Doug
Thanks, Doug. I too have it there
Hello,
Still about CDR and MySQL table, should the calldate field be inserted by
Asterisk?
This is the table structure we are using, based on Asterisk wiki:
mysql describe cdr;
+-+---+--+-+-++
| Field | Type
Hello,
I notice that it takes 4 to 6 seconds between someone pressing a cipher
and Asterisk continuing inside the dialplan. How come ???
Taken from verbose logfile :
(attempt 1)
[Jun 11 15:29:25] DTMF[18549] channel.c: DTMF begin '1' received on
SIP/SipAgenT01-1eb0
[Jun 11 15:29:25]
jg
Sent: Tuesday, June 11, 2013 5:28 AM
Playing an announcement like Your call has been... to A after C has
accepted the call is probably not a good idea, because C has to wait until
the the announcement has finished. In environments where callers are
announced to C, C would typically not want to
Are you using cdr_adaptive_odbc.conf to populate it? If so, there is no
Asterisk analog to calldate. You would need an alias set up. Mine looks
like:
alias start = calldate
so that the start of my call is what gets logged to the database as the
calldate.
Kevin Larsen
From: Jairo
Jonas Kellens wrote:
I notice that it takes 4 to 6 seconds between someone pressing a cipher and
Asterisk continuing inside the dialplan. How come ???
...
Why doesn't Asterisk continue immediately inside the dialplan after having
received the DTMF-input ?
Jonas,
Please provide the
On Tue, Jun 11, 2013 at 8:32 AM, Mordechay Kaganer mkaga...@gmail.comwrote:
B.H.
Hello!
We have several Asterik boxes that are connected to PSTN using PRI cards
and they are interconnected using IAX2 trunks so that incoming calls are
delivered from PSTN to the servers they belong to.
In
On 06/11/2013 04:12 PM, Matthew J. Roth wrote:
Jonas Kellens wrote:
I notice that it takes 4 to 6 seconds between someone pressing a cipher and
Asterisk continuing inside the dialplan. How come ???
...
Why doesn't Asterisk continue immediately inside the dialplan after having
received the
On Tue, Jun 11, 2013 at 9:29 AM, Jonas Kellens jonas.kell...@telenet.bewrote:
On 06/11/2013 04:12 PM, Matthew J. Roth wrote:
Jonas Kellens wrote:
I notice that it takes 4 to 6 seconds between someone pressing a cipher and
Asterisk continuing inside the dialplan. How come ???
...
Why
Yes, using cdr_adaptive_odbc.conf.
As it is a new table, just changed the name from calldate to start and now
it is inserting the field ok.
Thank you very much for your help.
Best.
2013/6/11 Kevin Larsen kevin.lar...@pioneerballoon.com
Are you using cdr_adaptive_odbc.conf to populate it? If
On 06/11/2013 04:39 PM, Richard Mudgett wrote:
On Tue, Jun 11, 2013 at 9:29 AM, Jonas Kellens
jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote:
On 06/11/2013 04:12 PM, Matthew J. Roth wrote:
Jonas Kellens wrote:
I notice that it takes 4 to 6 seconds between
The only way to resolve this is to redesign your dialplan so you do not have
ambiguous matching, This is not an Asterisk issue, this is an issue with the
way you designed your dialplan and would apply to any IVR on any system.
-Original Message-
From:
On 06/11/2013 04:44 PM, Jonas Kellens wrote:
[snip]
Ok thanks.
Any idea how I can resolve this ?
Even if there *can* be more than 1 digit, in case there is only 1 digit
it should go faster.
Would it help if they pressed for example 1 followed by the # key?
If not then, as Eric mentioned,
No. When you dial 1 the PBX does not know if it needs to match _X or _X.
-Original Message-
From: Jonas Kellens [mailto:jonas.kell...@telenet.be]
Sent: Tuesday, June 11, 2013 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Eric Wieling
Subject: Re:
On 06/11/2013 04:46 PM, Eric Wieling wrote:
The only way to resolve this is to redesign your dialplan so you do not have
ambiguous matching, This is not an Asterisk issue, this is an issue with the
way you designed your dialplan and would apply to any IVR on any system.
I understand that I
On Mon, Jun 10, 2013 at 04:06:27PM -0400, D'Arcy J.M. Cain wrote:
I am trying to build Asterisk on a NetBSD system but I am running into
two problems. The first only happens on an installation built from
NetBSD HEAD. The config variable HAVE_NEWLOCALE is erroneously set
during configure but
On Mon, Jun 10, 2013 at 04:10:23PM -0400, D'Arcy J.M. Cain wrote:
This is the second issue I found while trying to install Asterisk on a
NetBSD box. I can't load the rtp module because HAVE_OPENSSL_SRTP
seems to be set. Is there some way to simply force this variab;e to be
unset from a
Jonas Kellens wrote:
Even if there *can* be more than 1 digit, in case there is only 1 digit it
should go faster.
Jonas,
Use the TIMEOUT function to set the maximum amount of time permitted between
digits when the user is typing in DTMF. As you've discovered, the default is 5
seconds.
On Tue, 11 Jun 2013 18:42:07 +0300
Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Mon, Jun 10, 2013 at 04:06:27PM -0400, D'Arcy J.M. Cain wrote:
I am trying to build Asterisk on a NetBSD system but I am running
into two problems. The first only happens on an installation built
from
So, B transfers the call and after bridging to C, B should get an
announcement.
This is just an idea:
See whether you can dispatch the termination of the call leg B-C by
evaluating the DIALSTATUS variable. I am not sure whether you can see
this inside the dialplan, but you should get the
On Tue, 11 Jun 2013 18:43:55 +0300
Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Mon, Jun 10, 2013 at 04:10:23PM -0400, D'Arcy J.M. Cain wrote:
This is the second issue I found while trying to install Asterisk
on a NetBSD box. I can't load the rtp module because
HAVE_OPENSSL_SRTP seems
Since Dial() might not return, DIALSTATUS cannot be used.
I checked the various AMI events and you'll see a bunch of Newchannel,
Hangup, Bridge, Unlink, and Masquerade events when transferring calls.
You could use this to originate a call with the announcement for B. This
is ugly, but if B's
I need to install cdr_mysql.so module for logging call to mysql. I have
the source file cdr_mysql.c only. Can someone explain the steps needed to
get this module compiled and working in Asterisk 1.8.22.0 on CentOS.
Thanks.
Nick
--
How about
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DB.html
?
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
Hello Adam,
Thank you very much for your info.
Regards,
Jonson.
On Tue, Jun 11, 2013 at 12:34 AM, ad...@3a.hu wrote:
Hi,
On 06/10/2013 22:26, Jonson Player wrote:
Some users of main use + instead of 00 for international dial. Is there
any solution for this problem?
swap the + sign to
B.H.
On Jun 11, 2013 5:15 PM, Steve Totaro stot...@totarotechnologies.com
wrote:
On Tue, Jun 11, 2013 at 8:32 AM, Mordechay Kaganer mkaga...@gmail.com
wrote:
B.H.
Hello!
We have several Asterik boxes that are connected to PSTN using PRI cards
and they are interconnected using IAX2
hi,
I've solved various iax2 problem mentioning calltoken when I put
these lines in the iax configuration:
requirecalltoken=no
calltokenoptional=0.0.0.0/0.0.0.0
bye
Il 11/06/2013 19:25, Mordechay Kaganer
scrisse:
B.H.
32 matches
Mail list logo
