On 16/04/13 14:17, Gertjan Baarda wrote:
On 16 apr. 2013, at 15:08, Sebastian Arcus wrote:
I would like to access a Postgresql database directly from my dialplan (to
lookup names based on callerid numbers for incoming calls). Based on everywhere
I looked - it seems the only way to do this
I would like to access a Postgresql database directly from my dialplan
(to lookup names based on callerid numbers for incoming calls). Based on
everywhere I looked - it seems the only way to do this is with
func_odbc. Considering that Asterisk seems to be able to access
Postgresql databases dir
On 01/02/13 09:43, Hans Witvliet wrote:
-Original Message-
From: Olivier
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] OT - Chan-mobile -Bluetooth dongle on remote
LAN workstation
On 31/01/13 10:15, Olivier wrote:
2013/1/31 Sebastian Arcus mailto:s...@open-t.co.uk>>
On 31/01/13 07:25, Olivier wrote:
Hello,
On a LAN, is it possible to install a bluetooth dongle on one
workstation (at this time, this workstation OS is not spe
On 31/01/13 07:25, Olivier wrote:
Hello,
On a LAN, is it possible to install a bluetooth dongle on one
workstation (at this time, this workstation OS is not specified) and use
it with chan_mobile ?
I've read some USB over IP (or Ethernet) middleware exist but I'm not
certain I'm looking at the r
On 25/01/13 12:31, Johan Wilfer wrote:
2013-01-23 18:20, Sebastian Arcus skrev:
I have an Asterisk server with one SIP trunk to a SIP provider. As my
server registers with the SIP provider, I don't have any SIP ports open
at my end to the Internet. However, I have the RTP ports open (as SI
On 23/01/13 17:33, Carlos Alvarez wrote:
On Wed, Jan 23, 2013 at 10:20 AM, Sebastian Arcus mailto:s...@open-t.co.uk>> wrote:
I have an Asterisk server with one SIP trunk to a SIP provider. As
my server registers with the SIP provider, I don't have any SIP
ports open at my
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
Arcus
Sent: Wednesday, January 23, 2013 11:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Is there a need to secure RTP ports?
I h
I have an Asterisk server with one SIP trunk to a SIP provider. As my
server registers with the SIP provider, I don't have any SIP ports open
at my end to the Internet. However, I have the RTP ports open (as SIP
has some trouble with my NAT). My question is - what are the
vulnerabilities in thi
Behalf Of Patrick Lists
Sent: 16 October 2012 12:30 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] B200p card - use dahdi or mISDN?
On 10/16/2012 08:50 AM, Sebastian Arcus wrote:
I've just bought an OpenVOX B200p ISDN card - and if I remember
correctly from last time I use
On 16/10/12 11:30, Patrick Lists wrote:
On 10/16/2012 08:50 AM, Sebastian Arcus wrote:
I've just bought an OpenVOX B200p ISDN card - and if I remember
correctly from last time I used one of these - it is possible to use
either DAHDI or mISDN with it. Are there any factors to consider
I've just bought an OpenVOX B200p ISDN card - and if I remember
correctly from last time I used one of these - it is possible to use
either DAHDI or mISDN with it. Are there any factors to consider when
choosing which software to use? Is one better than the other - or does
one have features whi
Hi Hans,
On 18/09/12 08:04, Hans Witvliet wrote:
Hi all,
In one of my other project i had a look at chan_mobile.
I build 1.8.15.1 with the apropiate module. (in my distro asterisk is
build without chan_mobile ;-)
After i filled in the mac-addresses of the BT-adapter and the one from
my phone,
On 13/09/12 11:16, Olivier wrote:
2012/9/13 Benedikt Schöffmann mailto:benedikt.schoeffm...@gmail.com>>
Hi there,
I'm setting up a Asterisk network and I ran into some problems ...
as you might have guessed :)
The set up is like this:
Internal Communication in the compan
On 13/09/12 00:47, Vladimir Mikhelson wrote:
On 9/12/2012 5:33 PM, Sebastian Arcus wrote:
On 10/08/12 18:38, Chad Wallace wrote:
On Tue, 31 Jul 2012 09:44:26 +0100
Sebastian Arcus wrote:
I have two setups with SIP hardware phones as extensions and POTS
lines as trunks. Internal SIP to SIP
On 10/08/12 18:38, Chad Wallace wrote:
On Tue, 31 Jul 2012 09:44:26 +0100
Sebastian Arcus wrote:
I have two setups with SIP hardware phones as extensions and POTS
lines as trunks. Internal SIP to SIP calls are crystal clear, but all
calls bridged to POTS have a significant amount of static
I have two setups with SIP hardware phones as extensions and POTS lines
as trunks. Internal SIP to SIP calls are crystal clear, but all calls
bridged to POTS have a significant amount of static noise. The problem
is that if I plug a POTS phone directly into the line, there is almost
no static n
On 01/03/12 10:05, Sebastian Arcus wrote:
I have a server with an OpenVox A400P card with 2 FXO modules on it. The
internal extensions are SIP Grandstream phones. When making or receiving
external calls through PSTN, there is an interrupted hissing like high
pitch noise - which might go away for
On 11/03/12 18:57, Shaun Ruffell wrote:
On Sun, Mar 11, 2012 at 06:49:01PM +, Sebastian Arcus wrote:
On 11/03/12 14:07, Tzafrir Cohen wrote:
On Sun, Mar 11, 2012 at 12:05:48PM +, Sebastian Arcus wrote:
Hi all,
I've tried to explicitly set my two PSTN trunks/FXO lines to alaw
On 11/03/12 14:07, Tzafrir Cohen wrote:
On Sun, Mar 11, 2012 at 12:05:48PM +, Sebastian Arcus wrote:
Hi all,
I've tried to explicitly set my two PSTN trunks/FXO lines to alaw with:
alaw = 1-2
in /etc/dahdi/system.conf. However, when I do this, all I get is
loud intense noise on the
Hi all,
I've tried to explicitly set my two PSTN trunks/FXO lines to alaw with:
alaw = 1-2
in /etc/dahdi/system.conf. However, when I do this, all I get is loud
intense noise on the line and nothing else - can't dial, can't make
calls, can't receive calls. If I omit it altogether, everything
Hi,
I've tried both an AMD and an Intel motherboard - with identical results.
Sebastian
On 04/03/12 15:32, Carlos Rojas wrote:
Hello
Are you using a amd server?
Sometimes openvox doesn't work fine with amd processor
Regards
On Mar 1, 2012 2:07 PM, "Dave Platt" mailto:dpl...@radagast.org>>
On 01/03/12 19:07, Dave Platt wrote:
5. Placing ferrite cores on the phone cables.
Do either of the phone lines in question have DSL on them?
If so, a ferrite core (which will block common-mode RF
signals) probably won't help much, if at all. DSL is a
differential-mode signal, and its frequen
I have a server with an OpenVox A400P card with 2 FXO modules on it. The
internal extensions are SIP Grandstream phones. When making or receiving
external calls through PSTN, there is an interrupted hissing like high
pitch noise - which might go away for few seconds then start again.
1. The no
On 27/02/12 00:00, Gaurav P wrote:
Hi All,
I am using an Obi110 to bridge my PSTN line to Asterisk. Inbound and
outbound calls work fine, but I noticed that phones connected directly
to the PSTN line stop ringing as soon as Asterisk answers and rings one
of my extensions. I'd like the regular ph
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian Arcus
Sent: Sunday, February 26, 2012 5:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Possible bug (or feature?) in extension matching and
par
I wanted a custom extension to match miss-dialled numbers in my
dialplan. I've included the following:
exten => _X.,1,Answer()
exten =>
_X.,n,Playback(extension_not_found_please_make_sure_you_dial_nine_in_front_of_external_numbers)
exten => _X.,n,Hangup()
However, this has the curious side ef
On 22/11/11 23:22, Sebastian Arcus wrote:
Hi all,
I've been using a Nokia series 40 phone to receive GSM incoming
calls OK into Asterisk for a few years now. According to the
documentation of chan_mobile, it seems that (at least
Hi all,
I've been using a Nokia series 40 phone to receive GSM incoming
calls OK into Asterisk for a few years now. According to the
documentation of chan_mobile, it seems that (at least some) Nokia
mobiles with S60 operating systems should be able to also
On 31/10/11 14:14, Raj Mathur (राज माथुर) wrote:
On Monday 31 Oct 2011, Sebastian Arcus wrote:
Every time I start Asterisk (just by issuing /usr/sbin/asterisk),
the bash console text turns white. I'm using rxvt, so this makes
everything pretty much invi
Hello all,
I've googled around, and I've discovered that there was a bug
which turned the console text white when issuing help for certain
commands in the *Asterisk* console. That seems different from what
I experience. And that bug was fixed.
On 04/10/11 10:21, � wrote:
Am 04.10.2011 10:33, schrieb Sebastian Arcus:
Hello list,
I use Asterisk with one sipgate.co.uk trunk. Asterisk connects to
sipgate.co.uk as a sip agent/client (with "register =>" statement in
sip.conf).
If I restrict the number of ports used in
Hello list,
I use Asterisk with one sipgate.co.uk trunk. Asterisk connects to
sipgate.co.uk as a sip agent/client (with "register =>" statement in
sip.conf).
If I restrict the number of ports used in rtp.conf (to 1-10005 for
example) - will that affect the sip sessions to sipgate.co.uk a
On 02/10/11 21:58, dotnetdub wrote:
On 2 October 2011 21:36, Sebastian Arcus mailto:s...@open-t.co.uk>> wrote:
Just a follow up. I've opened up udp ports 1-2 on the Linux
box (where Asterisk is) and now I have sound. However, bear in mind
that the Netgear r
On 02/10/11 21:11, Sebastian Arcus wrote:
Everything is working fine, except bridging between the sipgate and
voipcheap trunks. I'll explain:
SNIP
What kind of firewall are you using?
SNIP
Just a follow up. I've opened up udp ports 1-2 on the Linux box
(where Asteri
oth trunks
are using the same codec. make sure you have the correct ports open.
make sure you force all udp traffic to flow through your astrisk switch
as well.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
---
On 02/10/11 20:42, dotnetdub wrote:
On 2 October 2011 16:20, Sebastian Arcus mailto:s...@open-t.co.uk>> wrote:
Hello list,
My setup is as follows:
Trunks: 2 sip trunks, one with voipcheap.co.uk
<http://voipcheap.co.uk>, one with sipgate.co.uk <http:
Hello list,
My setup is as follows:
Trunks: 2 sip trunks, one with voipcheap.co.uk, one with sipgate.co.uk
Extensions: 1 hardware sip phone
Asterisk: 1.8.7.0
Everything is working fine, except bridging between the sipgate and
voipcheap trunks. I'll explain:
1. If I call from an external phon
I'm thinking of implementing some easier to remember key sequences in
features.conf. Something like:
pickupexten = ##
atxfer => **
Can anybody think why this might be a bad idea, compared to the defaults
*8 and *2? I have made sure that no other feature uses single * or #, to
avoid matching
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Sebastian Arcus
Sent: Tuesday, 21 June 2011 11:10 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] "pickupsound = beep" kills call
pickup i
I have discovered that if I enable pickupsound = beep in features.conf,
if I try to do a pickup with *8, the calling channel keeps on ringing,
while the phone where I pick-up from shows that the call has been
answered (I don't know where though). Also, it seems to completely
bugger up my outgoi
ason for this? Could it be something I'm
doing - or should I report it as a bug?
Sebastian
On 20/06/11 19:00, Sebastian Arcus wrote:
I have problems using the call pickup under Asterisk 1.8.4.2. I have
another Asterisk with 1.6 - and it is working fine with the same settings.
I have setup the
I have problems using the call pickup under Asterisk 1.8.4.2. I have
another Asterisk with 1.6 - and it is working fine with the same settings.
I have setup the same callgroup and pickupgroup for all extensions in
sip.conf - just to make things simple for testing. The sequence *8 seems
to be c
On 16/06/11 19:12, Cassius Smith wrote:
Hello,
I do not use the skinny firmware. By the way, questions like this are best
shared with the asterisk-users group mailing list, so that a large segment
of the Asterisk community can benefit from the questions and answers.
Cassius Smith
Agreed
--
Hi,
Asterisk: 1.8.4.2
I've just managed to configure attended transfers using Asterisk and
Grandstream GXP-2000 phones. The only way I've got it to work is by
using one of the out-of-band DTMF modes on the phone (either RFC or
SIP-info).
I think I can understand why - as Asterisk wouldn't b
On 01/06/11 16:11, Sebastian Arcus wrote:
Asterisk - 1.8.4.1
Dahdi-linux - 2.4.1.2
Dahdi-tools - 2.4.1
Kernel: 2.6.37.6
Kernel BKL: enabled
I am upgrading Asterisk on this box. It has an OpenVox A400P PCI analog
card with 1 FXO and 1FXS module.
This server has been running just fine for two
Asterisk - 1.8.4.1
Dahdi-linux - 2.4.1.2
Dahdi-tools - 2.4.1
Kernel: 2.6.37.6
Kernel BKL: enabled
I am upgrading Asterisk on this box. It has an OpenVox A400P PCI analog
card with 1 FXO and 1FXS module.
This server has been running just fine for two years with Asterisk 1.6.1.0
I've just upgra
Hi,
On 05/09/2011 09:40 PM, Justin Sherrill wrote:
Anyone have some recommended equipment for alerting people to calls in a noisy
environment?
I have Polycom IP550 phones set up in some really noisy environments - our mine
hoists - and they tend to drown out the ringers. I'm using Clarity WR
On 05/09/2011 04:50 PM, Warren Selby wrote:
On Mon, May 9, 2011 at 10:48 AM, Sebastian Arcus mailto:s...@open-t.co.uk>> wrote:
That's strange. Mine get stuck on the booting phase, looking for the
tftp server, if they can't find it there. Even if I change the dhcpd
On 05/09/2011 03:40 PM, Warren Selby wrote:
Thanks for the reply. No, I run tftpd directly from rc.local script
(on Slackware). That's fine - I just wanted to know I wasn't doing
something wrong. If everybody else is in the same boat - I'll just
be along for the ride then :D
On 05/09/2011 08:47 AM, Jay R. Worthington wrote:
Hi,
i have some spare (read: Boss get's a new one every few month ;))
Android Phones laying around. Does someone know a way of using them as a
mobile gateway for asterisk? I could not find any SIP-Gateway in the
Market, and i don't think it's p
On 05/09/2011 12:02 PM, Doug Lytle wrote:
Sebastian Arcus wrote:
Cisco phones (at least the 7940) are supposed to be run with a tftp
server available at all time
That is my experience. But, if you're running tftp under Linux, then
it's probably spawned by xinetd and won't be
tftp setup
indispensable :)
On Sun, 2011-05-08 at 22:37 +0100, Sebastian Arcus wrote:
Hi James,
Thanks for the reply. I'm not concerned about performance. But I've
learned that every extra daemon software on a server comes with its
security caveats. I would feel much better about not h
rick Henry Hughes - 2009
On 5/8/2011 5:19 PM, Sebastian Arcus wrote:
Hi all,
Sorry for posting here - but I figured there are many people with
Cisco IP phones here - and I use them with Asterisk :-)
I have a couple of Cisco 7940 phones. I've loaded the SIP firmware OK,
loaded the SIP configu
Hi all,
Sorry for posting here - but I figured there are many people with Cisco
IP phones here - and I use them with Asterisk :-)
I have a couple of Cisco 7940 phones. I've loaded the SIP firmware OK,
loaded the SIP configuration files OK, they work with Asterisk just fine.
My question is -
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