That was it! I had a value (412 and 413) set for each phone. This overwrote the
caller ID that I was setting in the dialplan. Removing the contents of the
fromuser field cleared this issue.
Thanks Olle!
Brett Woollum
br...@woollum.com
- Original Message -
From: Olle E.
Thanks Danny.
I searched on internet and found a link :
http://forums.digium.com/viewtopic.php?f=1t=11135start=0
This link guided to use ODBC to store cdr into Oracle database but not
successful . Can you help me ?
Thanks and best regards.
On Tue, Nov 9, 2010 at 9:27 PM, Danny Nicholas
Hi
I am using asterisk 1.6.1.1 ,and trying to do cdr billing.But the
problem is when I do transfers,callforward,callparking cdr records are
not proper.Below is example
'A' called pstn number and 'A' transfer the call to 'B',In this
case cdr is genrating as
1 . caller - A
I updated an AsteriskNow system to 2.6.18-194.26.1.el5 with yum update.
Upon reboot dahdi modules cannot be found. In yum.log I see that
kmod-mISDN, kmod-dahdi-linux and kmod-dahdi-linux-fwload-vpmadt032 were
all deleted during the update.
I reinstalled the deleted packages but the dahdi modules
On 11/10/2010 06:55 AM, Frank Tarczynski wrote:
I updated an AsteriskNow system to 2.6.18-194.26.1.el5 with yum update.
Upon reboot dahdi modules cannot be found. In yum.log I see that
kmod-mISDN, kmod-dahdi-linux and kmod-dahdi-linux-fwload-vpmadt032 were
all deleted during the update.
I
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phuong Hoang
Sent: Wednesday, November 10, 2010 3:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Store CDR (call detail record) to
Hi list,
I'm trying to setup an asterisk 1.8.0 with MFC/R2, but I'm having these
error messages loading chan_dahdi:
module load chan_dahdi.so
ERROR[9241]: chan_dahdi.c:11848 mkintf: Signalling requested on channel
1 is MFC/R2 but line is in ISDN PRI signalling
ERROR[9241]: chan_dahdi.c:16180
Topic: Phono and Phono SDK, an exciting development you can read about
at http://phono.com
You can learn all about the technical aspects of Phono and the SDK
Friday with Chris Matthieu, but here are a couple of interesting
implementations that don't require much effort to show a
proof-of-concept:
Hello list,
I'm having some issues with some phones that don't stop ringing after
the call is answered somewhere else.
Basically, a call comes, enters a queue and all the phones in the queue
ring. The problem is that when the call is answered, some phones don't
stop ringing.
I don't know if it
On Wed, Nov 10, 2010 at 11:42:18AM -0300, Martin Spinassi wrote:
Hi list,
I'm trying to setup an asterisk 1.8.0 with MFC/R2, but I'm having these
error messages loading chan_dahdi:
module load chan_dahdi.so
ERROR[9241]: chan_dahdi.c:11848 mkintf: Signalling requested on channel
1 is
Hi
I have configured IAX2 realtime in Asterisk 1.6.2.13.
when I cannect a client to realtime extension, always the state of extension
is UNKNOW like:
* Name : marco
Secret : Set
Context : phones
Parking lot :
Mailbox : 2...@default
Dynamic : Yes
On Wed, 2010-11-10 at 17:10 +0200, Tzafrir Cohen wrote:
On Wed, Nov 10, 2010 at 11:42:18AM -0300, Martin Spinassi wrote:
Hi list,
I'm trying to setup an asterisk 1.8.0 with MFC/R2, but I'm having these
error messages loading chan_dahdi:
module load chan_dahdi.so
ERROR[9241]:
On 11/10/2010 09:31 AM, Martin Spinassi wrote:
On Wed, 2010-11-10 at 17:10 +0200, Tzafrir Cohen wrote:
On Wed, Nov 10, 2010 at 11:42:18AM -0300, Martin Spinassi wrote:
Hi list,
I'm trying to setup an asterisk 1.8.0 with MFC/R2, but I'm having these
error messages loading chan_dahdi:
module
El 10/11/10 10:31, Martin Spinassi escribió:
On Wed, 2010-11-10 at 17:10 +0200, Tzafrir Cohen wrote:
On Wed, Nov 10, 2010 at 11:42:18AM -0300, Martin Spinassi wrote:
Hi list,
I'm trying to setup an asterisk 1.8.0 with MFC/R2, but I'm having these
error messages loading chan_dahdi:
module
Thanks Danny.
Can you talk clearer about using ODBC to store cdr to oracle db because i
don`t want to synchronize oracle db with mysql db?
Thanks and best regards.
On Wed, Nov 10, 2010 at 9:15 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
On Wed, 2010-11-10 at 10:51 -0500, Miguel Molina wrote:
El 10/11/10 10:31, Martin Spinassi escribió:
On Wed, 2010-11-10 at 17:10 +0200, Tzafrir Cohen wrote:
On Wed, Nov 10, 2010 at 11:42:18AM -0300, Martin Spinassi wrote:
Hi list,
I'm trying to setup an asterisk 1.8.0 with MFC/R2, but
I'd love to, but I have bigger trees to chop in Asterisk that messing with
DB CDR's, so I just use the flat file /var/log/asterisk/cdr-csv/Master.csv.
I'll defer this answer to a poster who has actually climbed that mountain.
Apologies for terse answer and top-post.
_
From:
On Tue, 2010-11-09 at 17:38 -0800, Brett Woollum wrote:
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark
5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing
[...@sipphones:1] Set(SIP/413-0005, CALLERID(num)=2) in
new
Hi list,
how can I set up an peer, so that behind one IP (NAT) multiple devices
can access to this single peer to make outbound calls.
Some of these multiple devices will be SIP phones and these SIP phones
are trying to make registrations to this peer.
best regards
Thomas
--
Hello list,
I have an Asterisk setup with the following details:
1. 3 x internal extensions / sip hardphones - Grandstream 2000
2. 2 x internal extensions / dahdi cordless phone
3. 1 x 2 FSX ports OpenVOX pci card
4. 1 x internal sip extension / sip softphone (linphone)
5. 1 x 800Mhz Asterisk +
Dear All,
This question is related to an old zaptel version (1.2.14) that I'm using for
specific reasons on one of the machines in combination with Sangoma cards. The
version of zaptel/asterisk/libpri cannot be changed :-). Even though this is an
old version, I still hope somebody will be kind
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk
Sent: Wednesday, November 10, 2010 12:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] /dev/zap/channel
Hello List,
Is it possible to select ODBC_STORAGE without entering to 'menuselect'?
I'm currently building a package for my distro with a little script, and
would like to set this option without entering manually to 'menuselect'
I know that I could make the script to change the
Hi All,
I've got a realtime queue in place (strategy is wrandom), and have
added a member dynamically via queue add member . My agent shows in
the queue, but when he gets the call is not recognized as In Use.
Here is the output from queue show prior to the call:
*CLI queue show
QUEUE_3 has 0
On Wed, Nov 10, 2010 at 2:11 PM, Jose P. Espinal j...@slackware-es.com wrote:
Is it possible to select ODBC_STORAGE without entering to 'menuselect'?
$ ./configure
$ make menuselect.makeopts
$ menuselect/menuselect --enable ODBC_STORAGE menuselect.makeopts
$ make
...
--
Paul Belanger | dCAP
On Wed, 2010-11-10 at 11:09 -0800, Todd Fulton wrote:
Hi All,
I've got a realtime queue in place (strategy is wrandom), and have
added a member dynamically via queue add member . My agent shows in
the queue, but when he gets the call is not recognized as In Use.
Here is the output from
On Wed, Nov 10, 2010 at 2:42 AM, Olle E. Johansson o...@edvina.net wrote:
Any update, Paul?
Yes, still reproducible. I'm going to do some homework first before
creating a new issue on the tracker. Specifically, if anybody else
can reproduce it.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Thank you very much Paul.
Is that documented somewhere? (I could not find details in the ATFOT book)
Paul Belanger wrote:
On Wed, Nov 10, 2010 at 2:11 PM, Jose P. Espinal j...@slackware-es.com
wrote:
Is it possible to select ODBC_STORAGE without entering to 'menuselect'?
$ ./configure
$
Hi to all.
I have a problem in the AMI. Sometimes the AMI don't generate the event
NewState when the exten of destiny is Ringing and sometimes don't show me
the callerid in this events.
The event NewState what i refer:
Event: Newstate
Privilege: call,all
Channel: SIP/17-6fd6
ChannelState:
On Wed, Nov 10, 2010 at 07:55:54PM +0100, Asterisk wrote:
Dear All,
This question is related to an old zaptel version (1.2.14) that I'm using for
specific reasons on one of the machines in combination with Sangoma cards.
The version of zaptel/asterisk/libpri cannot be changed :-). Even
On Wed, Nov 10, 2010 at 3:01 PM, Jose P. Espinal j...@slackware-es.com wrote:
Is that documented somewhere? (I could not find details in the ATFOT book)
I'm not sure if it is. I've already add some notes to a wiki page [1]
I am working on.
[1]
On Wed, Nov 10, 2010 at 1:36 PM, Carlos Chavez cur...@telecomabmex.comwrote:
On Wed, 2010-11-10 at 11:09 -0800, Todd Fulton wrote:
Hi All,
I've got a realtime queue in place (strategy is wrandom), and have
added a member dynamically via queue add member . My agent shows in
the queue,
On Wed, 2010-11-10 at 08:38 +0100, Olle E. Johansson wrote:
6 nov 2010 kl. 15.30 skrev Hans Witvliet:
Hi all,
As stated in the subject, slightly off-topic, as it is not directly a
Asterisk issue, but more SIP in general
Because security in general, and specifically identification
Hey, all. I'm working on making a script to auto-provision my Polycoms.
I wanted one that:
- Gets the MAC by itself
- Fills in the provisioning info you supplied on a web page
- Creates appropriate files
- Reboots the phone (which then gets provisioned)
The last part was the sticking one,
Hi,
Setting callcounter = yes in sip.conf definitely made a difference! Now,
the agent state goes to Ringing. However, after the call is ended, the
state does not change out of Ringing, and new calls do not get routed to the
agent any longer. Below is after the 1st call has ended, and while a
Hi
Does anyone have the same problem, or know the solution?
Multiple warning messages on Asterisk 1.4.36: Dropping incompatible
voice frame on Local/
when receiving calls with codec A and doing multiple attended
transfers to codec B
Reproduced with the following channel combinations
SIP -
Hi Carlos.
Yes I did have fromuser set, which was the problem. I removed this for each
extension and that solved the issue.
Thanks!
Brett Woollum
br...@woollum.com
- Original Message -
From: Carlos Chavez cur...@telecomabmex.com
To: Asterisk Users Mailing List -
Thanks for your reply.now i decide to read from flat file
/var/log/asterisk/cdr-csv/Master.csv
and then insert into oracle database. it has been to meet you.
On Wed, Nov 10, 2010 at 11:46 PM, Danny Nicholas da...@debsinc.com wrote:
I’d love to, but I have bigger trees to chop in Asterisk that
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