Hello all,
Just wondering if anyone ever solved the Aastra 50-BLF limit when used
with Asterisk (any flavour)?
I know it's not strictly and Asterisk question - but I'm sure there's
plenty of you out there using Aastra's on the end.
Cheers,
Andrew Thomas
dCAP #1473
--
I have the following dialplan:
; calls prefix by '8' are recorded
exten = _8[01]./_251,1,Set(something=shortened)
exten = _8[01]./_251,n,Set(WAV=filename)
exten = _8[01]./_251,n,Monitor(wav,${WAV},mb)
exten = _8[01]./_251,n,Dial(mISDN/2/${EXTEN:1},,g)
exten =
On Wed, Feb 03, 2010 at 08:10:50PM +, Steve Howes wrote:
On 3 Feb 2010, at 19:17, John Regal wrote:
Can anyone tell me how I could originate a call from my server? My
use case is I am on the road and I want to dial into the server
using my cell phone, log into my user account and
Hello,
I am using 1.6.2.0 asterisk.
I wanted to use pickup by BLF button on asterisk..
But there is a little problem.
As i read on some pages, when BLF light blinking, if i press on that
extension, it sends ** to pickup that number.
I have
exten = _**.,1,Pickup(${EXTEN:2...@default)
exten =
Hi,
It seems to me that DECT Gigasets do not support MWI when connected to a
Polycom/Kirk DECT base station :
when a new message is dropped into user's mailbox, I can see a NOTIFY
message sent by Asterisk to Polycom/Kirk DECT base station (here a KWS300)
but Gigaset's handset MWI remains unlit.
Hi all,
OT but possibly of interest to many of you in the asterisk community,
Markus Feilner is our guest tomorrow on the VUC: VPN Users Conference.
Markus is an interesting guy. In a former life, Markus ran an asterisk
box and used Sipgate.de. He works for a German Linux publication and
just
Hello All,
Please let me know Answers to the following questions .Backgroud.
1. Which one is better to use libss7 or chan_ss7. Today first time i come to
know about it ... little bit i googled but need experts comment on it.
2. I have perpared my server ie installed Asterisk , Configured TE420P
On Thu, 4 Feb 2010, Per Jessen wrote:
; calls prefix by '8' are recorded
exten = _8[01]./_251,1,Set(something=shortened)
exten = _8[01]./_251,n,Set(WAV=filename)
exten = _8[01]./_251,n,Monitor(wav,${WAV},mb)
exten = _8[01]./_251,n,Dial(mISDN/2/${EXTEN:1},,g)
exten =
Steve Edwards wrote:
On Thu, 4 Feb 2010, Per Jessen wrote:
; calls prefix by '8' are recorded
exten = _8[01]./_251,1,Set(something=shortened)
exten = _8[01]./_251,n,Set(WAV=filename)
exten = _8[01]./_251,n,Monitor(wav,${WAV},mb)
exten = _8[01]./_251,n,Dial(mISDN/2/${EXTEN:1},,g)
exten =
Set the emailaddr into a channel variable. Since I'm there, just make your
h exten do the system if ${WAV} and ${emailaddr} are longer than 1. Like
this.
- exten = h,1,noop(hangup logic)
- exten = h,n,Gotoif($[${LEN(${WAV})} 4]?just_hangup)
- exten = h,n,Gotoif($[${LEN(${emailaddr})}
Hi Gang,
I'm working on a lumenvox app and am having fun with the
Gotoif's on speech/DTMF recognition. If you're using DTMF to enter a number
instead of speech to enter a numeric value, the engine will often return a
confidence score of 1000 instead of 1-999. Therefore this Gotoif
On Thursday 04 February 2010 09:05:36 Danny Nicholas wrote:
Set the emailaddr into a channel variable. Since I'm there, just make your
h exten do the system if ${WAV} and ${emailaddr} are longer than 1. Like
this.
- exten = h,1,noop(hangup logic)
- exten = h,n,Gotoif($[${LEN(${WAV})}
It's just come to my attention that newer phones from both Snom and
Grandstream support OpenVPN. Is this a new trend or something? Since
OpenVPN, in one swell foop, deals with both NAT issues and securing
communications, I'd be very interested in hearing if other phone vendors
were embracing
Danny Nicholas wrote:
Set the emailaddr into a channel variable. Since I'm there, just make
your h exten do the system if ${WAV} and ${emailaddr} are longer than
1. Like this.
- exten = h,1,noop(hangup logic)
- exten = h,n,Gotoif($[${LEN(${WAV})} 4]?just_hangup)
- exten =
It's actually worse than nonsensical if you're watching the CLI. It
generates a warning (and presumably wastes a bit of resources). Oh well,
that's why I don't work for Digium :)
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
- Ken D'Ambrosio k...@jots.org wrote:
It's just come to my attention that newer phones from both Snom and
Grandstream support OpenVPN. Is this a new trend or something?
Since
OpenVPN, in one swell foop, deals with both NAT issues and securing
communications, I'd be very interested in
Ken D'Ambrosio wrote:
It's just come to my attention that newer phones from both Snom and
Grandstream support OpenVPN. Is this a new trend or something? Since
Excellent! I'm glad you mentioned it. Now, I'm going to have to buy a
snom phone!
I think the quotes cause the values to be compared as strings, not
numbers. The old shell programmer's trick (which allows for empty
strings):
exten = s,n,GotoIf($[0${SPEECH_SCORE(0)} = 0${THRESHOLD}]?:tag)
ought to cause a numeric comparison.
--
Barry
On Thu, Feb 04, 2010 at 09:42:18AM
--
Antes de imprimir este mensaje piense bien si es necesario hacerlo:
El medio ambiente es cosa de todos.
--
AVISO LEGAL
Este mensaje, dirigido solamente a su destinatario, es confidencial.
Si lo ha recibido por error, CAJA DE GUADALAJARA le informa que su
contenido es reservado y no se
C
- Mensaje original -
De: Barry Miller asterisk-us...@notanet.net
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Enviado: Thu Feb 04 17:45:47 2010
Asunto: Re: [asterisk-users] Gotoif Question
I think the quotes cause the values to be
On Thu, 2010-02-04 at 09:55 +, Andrew Thomas wrote:
Hello all,
Just wondering if anyone ever solved the Aastra 50-BLF limit when used
with Asterisk (any flavour)?
I know it's not strictly and Asterisk question - but I'm sure there's
plenty of you out there using Aastra's on the end.
That's my dial plan lesson for today.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry Miller
Sent: Thursday, February 04, 2010 10:46 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Does anybody knows how to enable Stutter Tone on Audiocodes MP-114?
Similar feature like Sipura has, when message is left in a mailbox the unit
send a 0.5sec ring tone to the phone every 30min. or so.
IN:
SIP Advanced Parameters -- Supplementary Services:
Message Waiting Indication (MWI)
On Thu, Feb 4, 2010 at 11:30 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
- Ken D'Ambrosio k...@jots.org wrote:
It's just come to my attention that newer phones from both Snom and
Grandstream support OpenVPN. Is this a new trend or something?
Since
OpenVPN, in one swell foop, deals with
Hi All;
My Asterisk version is 1.4.19.1 and I am using the Pickup application, it works
when I try to pickup the call that was originated from extension (for example,
when 801 call 802, then the phone of extension 800 can pickup the call at 802).
But it does not work when someone call from
On Thu, 4 Feb 2010, Tilghman Lesher wrote:
On Thursday 04 February 2010 09:05:36 Danny Nicholas wrote:
Set the emailaddr into a channel variable. Since I'm there, just make your
h exten do the system if ${WAV} and ${emailaddr} are longer than 1. Like
this.
- exten = h,1,noop(hangup logic)
4 feb 2010 kl. 19.42 skrev Steve Totaro:
On Thu, Feb 4, 2010 at 11:30 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
- Ken D'Ambrosio k...@jots.org wrote:
It's just come to my attention that newer phones from both Snom and
Grandstream support OpenVPN. Is this a new trend or something?
On 02/04/2010 03:19 PM, Olle E. Johansson wrote:
Anyway - is there someone out there that know the behaviour of
OpenVPN in regardsof retransmits and such? A VPN that retransmits
will at some point hurt you if you transmit media over it, especially
if you scale it up.
Don't know the nuances,
I thought there was an option for this but cant find it
We have a busy callcenter and I would like the agents to log in and be
in a paused state upon login... Right now they login and they are
instantly receiving a call
Thanks for the input...
--
On Thu, February 4, 2010 3:19 pm, Olle E. Johansson wrote:
Anyway - is there someone out there that know the behaviour of OpenVPN in
regards of retransmits and such? A VPN that retransmits will at some
point hurt you if you transmit media over it, especially if you scale it
up.
OpenVPN
Just a thought; give them a penalty on login so unless the queue is full,
they will go a cycle without a call.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Grignon
Sent: Thursday, February 04, 2010
Olle E. Johansson wrote:
Anyway - is there someone out there that know the behaviour of OpenVPN in
regards of retransmits and such? A VPN that retransmits will at some point
hurt you if you transmit media over it, especially if you scale it up.
OpenVPN by default uses UDP, but can be
Alex Balashov wrote:
On 02/04/2010 03:19 PM, Olle E. Johansson wrote:
Anyway - is there someone out there that know the behaviour of
OpenVPN in regardsof retransmits and such? A VPN that retransmits
will at some point hurt you if you transmit media over it, especially
if you scale it up.
On 02/04/2010 03:48 PM, Doug Lytle wrote:
OpenVPN by default uses UDP, but can be configured to use TCP.
So, under UDP, there should be no issues with retransmits.
It does have a primitive built-in backward acknowledgment mechanism even
for UDP and lack of acknowledgment can cause
--[ UxBoD ]-- wrote:
Just taken delivery of a Snom 870 and one thing that did disappointment is
that you have to install a beta firmware to enable OpenVPN ... H ...
Please keep us informed on your thoughts about this phone. I'd like to
know before buying one for testing.
Doug
--
--[ UxBoD ]-- schrieb:
does anybody know how to reboot a SNOM M3 base station remotely ?
wget --user='admin' --password='admin' \
'http://snom-m3-ip-address/reboot.html'
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan
I've been waiting for this for years. Except that snom phones are crap -- I
would really like to see openvpn or ssh tunneling hacked into a Cisco phone...
But it's still awesome.
-dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On 02/04/2010 05:07 PM, David Gibbons wrote:
I would really like to see openvpn or ssh tunneling hacked into a Cisco
phone...
Not bloody likely.
--
Alex Balashov - Principal
Evariste Systems LLC
Tel: +1 678-954-0670
Direct : +1 678-954-0671
Web: http://www.evaristesys.com/
--
On Thursday 04 February 2010 14:09:39 Steve Edwards wrote:
On Thu, 4 Feb 2010, Tilghman Lesher wrote:
On Thursday 04 February 2010 09:05:36 Danny Nicholas wrote:
Set the emailaddr into a channel variable. Since I'm there, just make
your h exten do the system if ${WAV} and ${emailaddr} are
Per Jessen wrote:
I have the following dialplan:
; calls prefix by '8' are recorded
exten = _8[01]./_251,1,Set(something=shortened)
exten = _8[01]./_251,n,Set(WAV=filename)
exten = _8[01]./_251,n,Monitor(wav,${WAV},mb)
exten = _8[01]./_251,n,Dial(mISDN/2/${EXTEN:1},,g)
exten =
Why dont you use the MixMonitor application which allows for a system
command to be passed in as an argument that is executed once the
recording is finished??? -
MixMonitor(file.ext[|options[|command]])
command will be executed when the recording is over. Any strings
matching ^{X} will be
If call recordings were stored in stereo and the callers were evenly
distributed along the stereo spectrum. BAM.
Just a cool idea I thought up, but probably completely impossible, and
even if not, likely too much work for too little reward. Even less
likely would be live stereo conference
On Fri, 5 Feb 2010, Ben Dinnerville wrote:
Why dont you use the MixMonitor application which allows for a system
command to be passed in as an argument that is executed once the
recording is finished??? -
[snip]
Note that no environment variables are given to command — you must
pass these
I get a somewhat minimal set of standard shell environment variables
(BASH*, HOSTNAME, PWD, TERM, etc) including the same PATH environment
variable I passed to Asterisk when it was started.
That just means that you cant rely on environment variables in the
script that you execute and you
On Fri, 5 Feb 2010, Ben Dinnerville wrote:
(Referring to the environment variables available to the shell Asterisk
creates to execute the command specified in the mixmonitor() application.)
I get a somewhat minimal set of standard shell environment variables
(BASH*, HOSTNAME, PWD, TERM, etc)
Sorry it took awhile to answer.
DND works flawlessly, but whenever using BLF I can only tell that a line is
either in use (on a call) or not. I cannot tell a phone is on DND, or on hold
for that matter. Would be extremely useful.
Would be willing to pay for this developpement if it can
Hi,
I'm getting some strange behaviour on Asterisk 1.4 running on Debian
Stable (Lenny). I suspect it's something to do with my setup, rather than
a bug, but I'm struggling to see it, and would appreciate any input.
Setup: PC with two ethernet cards: eth0 goes to local network, including
two
I have various Cisco, Polycoms, Linksys, and SNOM phones. Only one
Snom but it is the 370 and I consider it a pretty good phone. I
haven't tested the bells and whistles, in fact, I personally, I am not
a bells and whistles kind of guy, but the phone is nice.
I bought a Snom many moons ago and
On 02/05/10 02:05, Nikhil Nair wrote:
Hi,
I'm getting some strange behaviour on Asterisk 1.4 running on Debian
Stable (Lenny). I suspect it's something to do with my setup, rather than
a bug, but I'm struggling to see it, and would appreciate any input.
Setup: PC with two ethernet cards: eth0
On Thu, 4 Feb 2010, Joseph wrote:
[...]
Does your router runs DHCPD, assigning network addresses on on your LAN?
Nope, the DHCPD on the ADSL router is disabled, because I'm running DHCPD
on the Debian box. In any case, the ADSL router is not directly
accessible from the local net.
If
On 02/05/10 02:35, Nikhil Nair wrote:
On Thu, 4 Feb 2010, Joseph wrote:
[...]
Does your router runs DHCPD, assigning network addresses on on your LAN?
Nope, the DHCPD on the ADSL router is disabled, because I'm running DHCPD
on the Debian box. In any case, the ADSL router is not directly
On 02/05/10 02:05, Nikhil Nair wrote:
Extract from sip.conf:
[general]
context=incoming
srvlookup=yes
realm=nikhil-nair.net
Your resolve authentication to an outside server, isn't it?
So here might be your problem; if there is no connection to the Internet no
authentication.
--
Joseph
--
On Thu, 4 Feb 2010, Joseph wrote:
On 02/05/10 02:35, Nikhil Nair wrote:
On Thu, 4 Feb 2010, Joseph wrote:
[...]
Does your router runs DHCPD, assigning network addresses on on your LAN?
Nope, the DHCPD on the ADSL router is disabled, because I'm running DHCPD
on the Debian box. In any
On Thu, 4 Feb 2010, Joseph wrote:
On 02/05/10 02:05, Nikhil Nair wrote:
Extract from sip.conf:
[general]
context=incoming
srvlookup=yes
realm=nikhil-nair.net
Your resolve authentication to an outside server, isn't it?
No, that's just a Realm string which has to match when the Asterisk
Hi Mike,
What version of spip are you using ?
Jimmy
-Original Message-From: l...@virtutel.caSent: Thu, 04 Feb 2010 20:46:07 -0500To: asterisk-users@lists.digium.comSubject: Re: [asterisk-users] Polycom phone DND state
Sorry it took awhile to answer.
DND works flawlessly, but
Hey Jimmy,
3.2.0 is what I am using.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Godbout
Sent: Thursday, February 04, 2010 22:21
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
You may be right.
Pressing DND will only return a BUSY dial status and so you really
cannot distinguish whether it is a genuine DND.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent:
On Thu, Feb 4, 2010 at 9:43 PM, Mike l...@virtutel.ca wrote:
*From:* l...@virtutel.ca
*Sent:* Thu, 04 Feb 2010 20:46:07 -0500
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] Polycom phone DND state
Sorry it took awhile to answer.
DND works flawlessly, but whenever
On Thu, Feb 4, 2010 at 9:20 PM, Nikhil Nair nn...@pobox.com wrote:
No, again, I can cut off the internet altogether with ifdown eth1, and
the SIP phones (via eth0) continue to work fine, as does the Zap channel.
It's only if eth1 is up but the ADSL router is down (or, indeed, the phone
line
Evenly distributed? like with conferences? or with Mixmonitor having
two sides to record?
On Thu, Feb 4, 2010 at 4:39 PM, Lyle Underwood lyleunderw...@gmail.com wrote:
If call recordings were stored in stereo and the callers were evenly
distributed along the stereo spectrum. BAM.
Just a cool
Hi Lyle,
What you are talking about is spatial distribution, I've already written
a post back in 2008 about it here;
http://blog.collins.net.pr/2008/08/diamondware-spatial-conferencing.html
Cheers,
Dean
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I have a Linksys Sipura SPA2102 connected to Asterisk 1.4.27 and
sometimes it doesn't connect at all. I keep getting a busy signal when
I try to dial.
It appears to happen most often when both lines are registered.
The 2 lines on Linksys lines also use different ports. Does that mean
than it is
On Thu, Feb 04, 2010 at 09:52:35PM -0600, Warren Selby wrote:
On Thu, Feb 4, 2010 at 9:20 PM, Nikhil Nair nn...@pobox.com wrote:
No, again, I can cut off the internet altogether with ifdown eth1, and
the SIP phones (via eth0) continue to work fine, as does the Zap channel.
It's only if
Oh cool. Now I'm actually starting to wonder how hard it would be to
implement this. Unfortunately I'm extremely new to Asterisk and have not
even considered development for it. Hell, I use Trixbox.
I started googling around though and I found out that FreeSwitch
actually already has the feature
On Thursday 04 February 2010 23:22:27 Alex Samad wrote:
What I have seen on my asterisk box when I had a up/down adsl line was
that the asterisk box couldn't do dns resolution and would hang( well no
other internal calls could be made, seemed like some sort of semaphore
was stuck) when the
I have very little idea of what any of this stuff means. I know very
little about the inner-working of Asterisk, and I sort of joined this
list to glean some knowledge. All I've heard about Mixmonitor is
somebody talking earlier about how it was called after a recording was
completed to handle
Anyway - is there someone out there that know the behaviour of OpenVPN in
regards of retransmits and such? A VPN that retransmits will at some point
hurt you if you transmit media over it, especially if you scale it up.
OpenVPN is well-behaved in that way. It uses SSL over TCP for its
On 1.6.2 I have also tried using a local channel for the outbound leg
with the originate looking like the following:
action:.Originate..
actionid:.1306903_89#AJ_ORIGINATE_25
timeout:.4
exten:.s
async:.true
callerid:..612
Ben Dinnerville wrote:
Why dont you use the MixMonitor application which allows for a system
command to be passed in as an argument that is executed once the
recording is finished??? -
MixMonitor(file.ext[|options[|command]])
command will be executed when the recording is over.
I did
4 feb 2010 kl. 21.54 skrev Alex Balashov:
On 02/04/2010 03:48 PM, Doug Lytle wrote:
OpenVPN by default uses UDP, but can be configured to use TCP.
So, under UDP, there should be no issues with retransmits.
It does have a primitive built-in backward acknowledgment mechanism even
for
5 feb 2010 kl. 06.49 skrev Anthony Messina:
On Thursday 04 February 2010 23:22:27 Alex Samad wrote:
What I have seen on my asterisk box when I had a up/down adsl line was
that the asterisk box couldn't do dns resolution and would hang( well no
other internal calls could be made, seemed like
On Fri, Feb 5, 2010 at 1:46 AM, Dave Platt dpl...@radagast.org wrote:
Anyway - is there someone out there that know the behaviour of OpenVPN in
regards of retransmits and such? A VPN that retransmits will at some point
hurt you if you transmit media over it, especially if you scale it up.
72 matches
Mail list logo
