I believe TIMESTAMP no longer works in 1.4. You need to use the below
statement or a variation on it. The documentation does include how to
use this.
{STRFTIME(${EPOCH},,%Y%m%d-%H:%M:%S)}
This will give you the time and date as 20070618-15:36:17.
You can place the variables in any order or forma
Not necessarily. If you set the following in the sip.cfg file to 1 it
will reboot even if there are no changes. The default is zero which will
only reboot if there are changes.
voIpProt.SIP.specialEvent.checkSync.alwaysReboot="1"/>
___
Kevin Savoy
Business Unit Telecom Analyst
2218
Not sure if this can be done or not, but I can't seem to find it
anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I
would like to have the caller id of the person I am dialing displayed
and not the number I just dialed. Is this possible? So, if extension
4023 is John Doe, and I d
We are looking at using Asterisk as a call recording server for an Avaya
VoIP S8700 system in a multi-site VoIP Call Center. All calls will be
coming in to one location and sent out via VoIP to other call centers.
What kind of specs should we be looking at purchasing for our Asterisk
server to
Savoy, Kevin - Williston, ND wrote:
> Well thanks to those who did reply. I guess I'll have to live with it
> until somehow it gets fixed. The reason I upgraded to 1.4 is that there
> were three or four other issues I had that this fixed. Going back just
> isn't really an opt
CTED] ' <-- replace the context with whatever
you are using.
Eric Osterberg
Sound Choice Communications LLC
- Minnesota, US
On Wed, 14 Feb 2007, Savoy, Kevin - Williston, ND wrote:
> I am having an issue with 1.4 where we can't successfully transfer a
> call dir
:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: FW: [asterisk-users] Problem Transferring Direct to
Voicemail
Maybe nobody knows. I certainty know that I've never ever seen that
error.
Savoy, Kevin - Williston, ND wrote:
> Could someone at least respond to
th nothing telling me what Notify answer on an owned
channel means and what to do about it.
PLEASE!! Someone?? Anyone???
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Savoy,
Kevin - Williston, ND
Sent: Wednesday, February 14, 2007 8:29 AM
To
I am having an issue with 1.4 where we can't successfully transfer a
call directly to a voicemail box. We hit "Transfer" on the phone and
dial the mailbox number we want to send it to,
My dial plan for this is:
exten=>_*40XX,n,Voicemail(${EXTEN:1},u)
The voicemail system picks up and st
No one knows what the Notify answer on an owned channel is?
Anyone?
-Original Message-
From: Savoy, Kevin - Williston, ND
Sent: Monday, February 12, 2007 11:14 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: FW: [asterisk-users] After upgrade to 1.4
Try "yum install gcc-c++"
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of younss
azzayani
Sent: Tuesday, February 13, 2007 8:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] error when compiling asterisk-1.4
Sounds like you don't have the gcc-c++ package installed.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Tuesday, February 13, 2007 6:53 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] error when compiling asteris
risk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] After upgrade to 1.4 transfers don't workproperly
On Wed, 2007-02-07 at 14:12 -0600, Savoy, Kevin - Williston, ND wrote:
> I have discovered an issue on my system after upgrading from 1.2.13 to
> 1.4. A call c
rs don't workproperly
On Wed, 2007-02-07 at 14:12 -0600, Savoy, Kevin - Williston, ND wrote:
> I have discovered an issue on my system after upgrading from 1.2.13 to
> 1.4. A call comes in on a T1 line and goes to a Polycom 501 SIP phone.
> I have confirmed this on multiple phones.
don't workproperly
On Wed, 2007-02-07 at 14:12 -0600, Savoy, Kevin - Williston, ND wrote:
> I have discovered an issue on my system after upgrading from 1.2.13 to
> 1.4. A call comes in on a T1 line and goes to a Polycom 501 SIP phone.
> I have confirmed this on multiple phones. W
I have discovered an issue on my system after upgrading from 1.2.13 to
1.4. A call comes in on a T1 line and goes to a Polycom 501 SIP phone. I
have confirmed this on multiple phones. When the called person answers
and tries to transfer the call to another extension, the call
successfully transfers
I find this surprising. Is this fact? I don't see faxing disappearing
anytime soon. I'm surprised Asterisk/Digium would ignore it and not try
to support it. If people are to replace their old PBX's with Asterisk
faxing is almost always going to be required. We have an old POTS line
for faxing but i
I had the same issue. I needed to install #yum install mysql-devel.
Once I did this the addons compiled the file fine.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pablo
Almido
Sent: Wednesday, January 17, 2007 9:43 AM
To: asterisk-users@
We have been running an Asterisk box with 1.2.9.1 on it since August in
a call center environment. We use the Asterisk box as an IVR and then
pass the calls on to a Nortel Option 11C. Today we found in our long
distance bill two calls that lasted a VERY long time. One was 58 hours
and another was 3
Is there a way to detect which end of a call hung up? If so can I log
this to the CDR records? Any pointers or can anyone point me to where I
can get this info?
_
Kevin Savoy
Business Unit Telecom Analyst
2218 4th Ave W
Williston, ND 58801
Ph: 701-774-4023
Fax: 701-
We tried them out early last year when we were looking at a large deployment
and they gave us a lot of the redundancy that we wanted. However we did run
into issues where calls seemed to get caught up in the system. It was as far as
we could tell rather random. No consistency to it at all. Aster
I had this same problem. It was that I was missing the mysql-devel
package. I installed this on my Fedora Core 4 system with "yum install
mysql-devel". Once I installed this I redid the ./configure, make and
make install of the addons and voila it was there.
Fr
, Kevin - Williston, ND wrote:
> Ok so I'm the only one not getting this to work. Maybe I'm doing
> something wrong. Here is the installation I'm using. Install Fedora
Core
> 4 and do all the updates through yum. Then I install zdlib-devel,
> openssl-devel, newt-devel, gcc,
ks
-Original Message-----
From: Savoy, Kevin - Williston, ND
Sent: Thursday, December 28, 2006 9:17 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: FW: [asterisk-users] cdr_addon_mysql.so did not register itself
duringload
So no one else is having issues w
So no one else is having issues with MySQL and 1.4? I'm the only one?
-Original Message-
From: Savoy, Kevin - Williston, ND
Sent: Wednesday, December 27, 2006 2:09 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] cdr_addon_m
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua
Colp
Sent: Tuesday, December 26, 2006 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] cdr_addon_mysql.so did not register itself
duringload
Savoy, Kevin - Williston, ND wrote:
>
>
> I've loaded
I've loaded Asterisk 1.4 with the addons 1.4, libpri 1.4 and Zaptel 1.4
as well. I can place calls but I noticed the MySQL was writing out to
the database. When doing an Asterisk load with asterisk - I saw the
following:
[Dec 26 11:02:08] WARNING[10029]: loader.c:375 load_dynamic_module:
Mo
I was able to get it to work with 2 extensions. One for the "host" and
one for the "participants" Not the best way to set it up but it works.
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Savoy,
Kevin - Williston, ND
Sent:
ling List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MeetMe Conferencing and Marked Mode
On 12/13/06, Savoy, Kevin - Williston, ND <[EMAIL PROTECTED]> wrote:
> I am trying to set up a Conference room where users are put on hold
> until the host arrives. I have figured out that the
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MeetMe Conferencing and Marked Mode
Am Dienstag, den 12.12.2006, 13:08 -0600 schrieb Savoy, Kevin -
Williston, ND:
> I am trying to set up a Conference room where users are put on hold
> until the host arrives. I have
Try using MixMonitor instead.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of nik600
Sent: Tuesday, December 12, 2006 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [BULK] [asterisk-users] Asterisk manager
Importance: Low
Hi
I am trying to set up a Conference room where users are put on hold
until the host arrives. I have figured out that the A option activates
marked mode and the w option is used to activate the waiting until the
marked user arrives. This seems to be what I need. What I can't seem to
find is how do I
That was it for me as well. Couldn't get that answer the last time I
asked. Thanks guys.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Tuesday, October 10, 2006 11:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
S
If you get an answer for this please post it here on forum as I and at
least one other I've talked to have this same problem. I found it was
only a problem from external calls though not internally. Same for you?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behal
Ronnie I have 4 non-PRI’s connected
to a Nortel 11C and I had played with PRI connections before and got them
working. If you want to call me we can go over your set up and compare with
mine.
Kevin Savoy
701-774-4023
Novo1
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
.
--
--
Steven
http://www.glimasoutheast.org
"Savoy, Kevin - Williston, ND" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]...
We
are trying to set up a script that will test hundreds of toll free numbers to
ensure that they correctly terminate a
We
are trying to set up a script that will test hundreds of toll free numbers to
ensure that they correctly terminate at our Nortel PBX. We have the perl
scripting written to dial the numbers and it works like a charm except for one
problem. We are not sure how to detect whether a successfu
Is there a way to contact her directly or do we have to go through
Digiums website?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid
Bender
Sent: Sunday, August 27, 2006 11:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [BU
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