...@lists.digium.com] *On Behalf Of *Bart
Coninckx
*Sent:* Tuesday, September 25, 2012 4:23 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] no audio while call forwarding, yes audio
with followme
Hi all,
the subject says it all.
Technical details
Hi all,
the subject says it all.
Technical details:
- Asterisk 1.8.7.1
- Behind NAT
- Using external SIP provider
The call forwarding is tested both with this functionality on the phone
and with configuration in the dialplan. In the latter case a database
variable is set to the external
On 05/10/12 00:09, Richard Mudgett wrote:
Please just reply to the mailing list.
oops, that was my intention, my bad.
Are you able to make calls when in PTP mode?
I just tested: yes it seams so!
The warning message is just
complaining about receiving unexpected TEI management messages
Hi all,
for smaller (or maybe even bigger) sites I'm looking for a smaller,
appliance-type like PC, preferably solid state and fanless PC.
Since it's only going to run Asterisk for a couple of extensions I don't
think CPU and RAM need to be maxed out.
Does anyone have inspiration/experience
On 05/10/12 13:49, A J Stiles wrote:
On Thursday 10 May 2012, Bart Coninckx wrote:
I'm looking for a smaller,
appliance-type like PC, preferably solid state and fanless PC.
Since it's only going to run Asterisk for a couple of extensions I don't
think CPU and RAM need to be maxed out.
Does
John Novack
Bart Coninckx wrote:
Hi all,
for smaller (or maybe even bigger) sites I'm looking for a smaller,
appliance-type like PC, preferably solid state and fanless PC.
Since it's only going to run Asterisk for a couple of extensions I
don't think CPU and RAM need to be maxed out.
Does
, but there are
Sockris and ALIX flash based boards. AstLinux has special configurations for
these.
I have 20-30 AstLinux on thin clients working without a belch on a private
collectors network
John Novack
Bart Coninckx wrote:
Hi all,
for smaller (or maybe even bigger) sites I'm looking
available Asterisk cluster (using DRBD and Pacemaker).
Anyone 2 cents about that?
BC
On 05/10/12 14:28, John Novack wrote:
Correct. I have never been accused of being a good speller!
JN
Bart Coninckx wrote:
That's Soekris I suppose. Never heard of them, but it looks mighty
interesting
PC suitable
for Asterisk
Correct. I have never been accused of being a good speller!
JN
Bart Coninckx wrote:
That's Soekris I suppose. Never heard of them, but it looks mighty
interesting.
Cheers,
BC
On 05/10/12 13:35, John Novack wrote:
I use HP Thin Clients with AstLinux installed.
HP
Tim,
looked at these briefly, they all seemed pre-installed, correct? Is
reinstallation with, let's say, CentOS possible?
thx,
BC
On 05/10/12 14:39, Tim Nelson wrote:
Bart Coninckx wrote:
Hi all,
for smaller (or maybe even bigger) sites I'm looking for a smaller,
appliance-type like PC
On 05/10/12 18:38, Kevin P. Fleming wrote:
On 05/10/2012 03:49 AM, Bart Coninckx wrote:
Hi all,
for smaller (or maybe even bigger) sites I'm looking for a smaller,
appliance-type like PC, preferably solid state and fanless PC.
Since it's only going to run Asterisk for a couple of extensions I
Hi,
I'm experiencing difficulties to get a B410P running with Asterisk
10.3.1 and DAHDI 2.6.1.
Am I supposed to use DAHDI for this card and ISDN BRI for my country
(Belgium)?
thx,
BC
--
_
-- Bandwidth and Colocation
it helps!
On Wed, May 9, 2012 at 1:59 PM, Bart Coninckx
bart.conin...@telenet.be mailto:bart.conin...@telenet.be wrote:
Hi,
I'm experiencing difficulties to get a B410P running with Asterisk
10.3.1 and DAHDI 2.6.1.
Am I supposed to use DAHDI for this card and ISDN BRI for my
Right you are,
but when using bri_cpe I get:
[May 9 23:08:45] WARNING[2775]: sig_pri.c:6969 pri_dchannel: PRI Error
on span 4: Received MDL/TEI managemement message, but configured for
mode other than PTMP!
This repeats itself every second.
The
bri_cpe_ptmp
settings seems to give the
All,
has anyone any experience in using Wifi smartphones as SIP clients? Does
this work properly? What models/brands are optimal for this (in terms of
ease of use, battery life etc)?
Thx!!
B.
--
_
-- Bandwidth and
.
Use good quality Access Points like Ruckus Wireless.
Mitul
On May 7, 2012 1:55 PM, Bart Coninckx bart.conin...@telenet.be
mailto:bart.conin...@telenet.be wrote:
All,
has anyone any experience in using Wifi smartphones as SIP
clients? Does this work properly? What models/brands
Benny,
very useful, thank you.
So, in short, at this stage it's best to go DECT for wireless and if
DECT and Wifi need to be combined (because both types of devices exist
in the organization), it's preferable to go to access points that offer
both networks.
Correct?
thx,
BC
On 05/07/12
What about phones like the Unidata WPU-7800 (
http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have
experience with those? Would these also suffer from connection losses?
thx,
BC
On 05/07/12 10:57, Benny Amorsen wrote:
Bart Coninckxbart.conin...@telenet.be writes:
has anyone
://www.entvoice.com/
email: mi...@enterux.in mailto:mi...@enterux.in
DID: +91-22-61447605
Cell: +91-9820332422
On Mon, May 7, 2012 at 3:33 PM, Bart Coninckx
bart.conin...@telenet.be mailto:bart.conin...@telenet.be wrote:
What about phones like the Unidata WPU-7800 (
http://www.udcsystems.com
On 05/07/12 13:04, Benny Amorsen wrote:
man, 07 05 2012 kl. 12:03 +0200, skrev Bart Coninckx:
What about phones like the Unidata WPU-7800
( http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have
experience with those? Would these also suffer from connection losses?
I don't know
On 05/07/12 13:50, giovanni.v wrote:
I thing smartfones simply lack a business grade softhone implementation.
WiFi SIP phones share some problem with smartphones: battery runtime
and, most important, roaming and handover.
In WiFi network handling roaming/handover is up to the client, this is
Hi,
Looking to do video conferencing with Asterisk and after some research I
noticed there's mainly the new confbridge application in Asterisk 10 or
there's the Medooze MCU software.
I'm not sure as how they compare feature-wise. I get the impression the
video support in confbridge is
Hi all,
After having done some successful tests with 3G in combination with the
G729a codec, I plan to use this as a failover path for when the main
internet connection goes down.
However, on this usual connection, G711a is used. I could have the
script that monitors the main line also sed
Hi all,
when doing a blind transfer using the keys defined in features.conf, we
hear a confirmation of the attempt to blindly transfer, followed by an
invalid extension message.
The console says this:
[Jun 4 22:30:31] VERBOSE[11301] res_musiconhold.c: -- Started music
on hold, class
Hi all,
I have a rather old 1.4 installation that recently was connected to a new
network via an IPSEC tunnel. No NAT-ing is involved anywhere (I've seen posts
about the same phenomenon but with NAT). It first the phones on the PBX
network did not get the audio of the phones on the remote
Hi,
I've built an Asterisk HA cluster by means of heartbeat and drbd. The
following folders are stored on shared storage and referred to by means of
symbolic links:
/etc/asterisk
/var/lib/asterisk
/usr/lib/asterisk
/var/spool/asterisk
/var/log/asterisk
I was under the impression that phone
Hi,
I'm using a ISDN-30 E1 line from KPN Belgium.
The challenge is to get a correct CallerID on outgoing lines.
When I put this in my dialplan:
exten = _0.,1,Set(TEMPVAR=${CALLERID(num):1})
exten = _0.,2,Set(CALLERID(num)=144622${TEMPVAR})
exten = _0.,3,NoOp(${CALLERID(num)})
exten =
Hi,
when I have this in my dialplan in order to get a cascade for incomming calls
exten = 611,1,Dial(SIP/611,10)
exten = 611,2,Dial(SIP/607,60)
exten = 611,3,Dial(SIP/620,60)
exten = 611,4,Hangup()
I get a ringing tone for the first Dial command, but the others produce
silence, even when I use
Hi all,
when enabling blind and attended transfers in features.conf, these only seem
to work when I enable voicemail for a particular user. How can this be? Can I
have transferrring without voicemail?
Using Asterisk 1.4 by the way.
Thank you!
Bart
Hi all,
when enabling blind and attended transfers in features.conf, these only seem
to work when I enable voicemail for a particular user. How can this be? Can
I
have transferrring without voicemail?
Using Asterisk 1.4 by the way.
Thank you!
Bart
I think some clarification
30 matches
Mail list logo
