Hi Jacek,
Thank you very much for the suggestion. Using SetVar and
CONNECTEDLINE(number) works.
On 12 December 2016 at 18:31, Jacek Konieczny <jaj...@jajcus.net> wrote:
> On 2016-12-12 02:21, David Cunningham wrote:
>
>> Is there any equivalent of the CONNECTEDLINE
Hello,
Is there any equivalent of the CONNECTEDLINE function which can be called
from an application using the AMI?
Thanks for any ideas.
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http://voisonics.com/
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(that they are sending the
DID number and what format it is in).
All the best,
David
On 27 Oct 2016 5:21 am, "KyD" <k...@lumac.com.ar> wrote:
Hi,
My sip provider gave me 2 numbers for the incoming call via pstn.
nro1 = 12341234
nro2 = 45674567
I have a dialplan for each.
if i put thi
ere:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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>
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Digium, Inc. · Dir
. Specialization is for insects.
>> ---Heinlein
>>
>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us f
at www.astricon.net
Hope I get to see you in Glendale!
All the best,
David
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Digium, Inc. · Director, Worldwide Asterisk Community
6 Landscape Close · Weston on the Green · Bicester · Oxfordshire OX25 3SX ·
UK
direct/fax: +1 256 428 6119 · mobile: +44 7722 442236
uct-phone-217b;2 Opened file 0
'/var/lib/product/music/2/2/1'
[Mar 10 08:00:40] DEBUG[3158][C-273f] res_timing_timerfd.c: Expected to
acknowledge 1 ticks but got 4 instead
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it seemed that the [asterisk-1] section in pjsip.conf had
no effect. Our sorcery.conf is attached.
Is this possible, and how do we do it? Thanks very much for any advice.
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so
Shame, but thank you very much for the reply Joshua.
On 22 January 2016 at 10:26, Joshua Colp <jc...@digium.com> wrote:
> David Cunningham wrote:
>
>> Hello,
>>
>> Is it possible to mix PJSIP realtime and flat file configuration in
>> pjsip,conf?
>&
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Hello,
Can anyone advise on the status of SRV lookups in Asterisk 11?
(specifically 11.17.1)
Is there any difference given how the Dial is done, and how supported are
weights and priorities?
Thanks in advance,
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- Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Asterisk 13 chan_sip trunk appending
@string to dialled number
David,
I should also note;
246 is my extension, it has IP 172.22.3.238.
172.22.4.8 is the PBX, and 172.22.4.12 is the 2800 gateway.
The trunk is called
/asterisk and grep for OUT_3_SUFFIX in all the
files
once the file with that variable is located, we can figure out why it's
adding it
On 08/17/2015 11:26 PM, David Cunningham wrote:
Yes indeed.
Do you have the dialplan, eg from /etc/asterisk/extensions.conf?
Something is getting
:
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by http://www.api-digital.com --
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webinar every Thurs:
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introductory webinar every Thurs:
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asterisk-users mailing list
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Digium, Inc. · Director, Worldwide Asterisk Community
The mediatrix 4102s line kicks ass.
On Jun 15, 2015 8:49 PM, Matt Darnell mattdarn...@gmail.com wrote:
In the past we have used Adtran Atlas 550's to break out FXS ports for
devices like modems. The great thing about the 550 is that internally it
is all TDM so there is absolutely zero
On some SIP phones it is possible to turn off the missed call
notifications, but I am not aware of a way to do the same on any cell
phones.
On 5 Jun 2015 07:29, Mehmet Avcioglu meh...@activecom.net wrote:
There is no signal that is sent to display a missed call. Your cell phone
does that. If
Remember that that zyxel 16 port switch is only 8 poe ports. If your phones
are 802.3af or 802.3at, you could look at the ubiquiti line of switches.
On Mar 13, 2015 9:34 PM, Brian Franklin bfrank...@ntginc.net wrote:
If your phones support PoE,
I have had huge success with Zyxel:
In a word, no.
PRI service providers will generally only allow the caller ID to be set to
one of the numbers in the range that you have for inbound with them.
On 18 Mar 2015 11:30, Rizwan H Qureshi rizwanhas...@gmail.com wrote:
Hi All,
I have to forward incoming call on PRI back out to PRI but
for any suggestions.
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New
coming out in May.. If they're any
good we'll strongly consider those...
dw
On Mon, Mar 9, 2015 at 10:55 PM, Ryan Wagoner rswago...@gmail.com wrote:
On Mon, Mar 9, 2015 at 9:40 AM, David Wessell da...@ringfree.biz wrote:
Welcome to our hell.
We ran into this on VVX 300 and 400 phones running UCS
Welcome to our hell.
We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We finally
got Polycom to issue a hotfix firmware version. I'll be happy to share it
with you offlist, just email me.
Officially Polycom will fix the issue in 5.3 in a few months..
Thanks
David
On Mon, Mar 9
I'll add that it appears to happen when you have users in a ring group or
call queue and BLF is being used in some capacity..
dw
On Mon, Mar 9, 2015 at 9:40 AM, David Wessell da...@ringfree.biz wrote:
Welcome to our hell.
We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We
*
This way, you will be setting the caller ID with a name label that can be
observed on the SIP client before answering.
All the best,
David
On 4 March 2015 at 11:53, Mark Rogers m...@more-solutions.co.uk wrote:
Background: I dabbled with asterisk years ago, and more recently have
more
itself, or in Asterisk.
Unfortunately, I doubt the Python test suite would run on non-Linux. I don’t
even bother trying to run it on Ubuntu; I have a CentOS VM specifically for
running the test suite to avoid platform problems.
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Digium, Inc. | Software Developer
445 Jan Davis Drive
On Feb 12, 2015, at 10:42 AM, Matthew Jordan mjor...@digium.com wrote:
On Thu, Feb 12, 2015 at 10:38 AM, David M. Lee d...@digium.com
mailto:d...@digium.com wrote:
Unfortunately, I doubt the Python test suite would run on non-Linux. I don’t
even bother trying to run it on Ubuntu; I
)
exten = _.,n,Set(TRANSFER_CONTEXT=transferencia)
exten = _.,n,Playback(tt-weasels)
exten = _.,n,Goto(2)
exten = _.,n,Hangup
Thank you in advance for your help
David Pinedo
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I can confirm that all the videos from AstriCon 2014 will be available at
www.AstriCon.net within about 3 weeks.
On 29 Oct 2014 16:33, Jeff LaCoursiere j...@jeff.net wrote:
On 10/29/2014 05:50 AM, Bogdan Cristea wrote:
Hi
Will the presentations made at Astricom 2014 be made public as
Hello,
Is there a way to tell in the dialplan, Asterisk AGI, or Asterisk AMI, when
a call has been hung up because the SIP rtptimeout has been reached?
Thank you,
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Step 1 would be an 'lspci' on the Linux command line to see if the Linux
box recognises the card
Step 2 would be to ensure that your DAHDI version is new enough to work
with the card
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[image: Digium logo]
*David
Tim,
I THINK but I'm not sure that you can do this with the Polycom multicast
page function. Have you attempted this yet?
Thanks
david
On Tue, Sep 16, 2014 at 10:07 PM, Tim Nelson tnel...@rockbochs.com wrote:
Greetings-
As many of your are Polycom experienced, I was hoping some kind soul
asterisk-users mailing list
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Great news!
On Wednesday, August 20, 2014, Roberto Fichera ker...@tekno-soft.it wrote:
On 08/04/2014 03:03 PM, David Duffett wrote:
Please come back to let us know if this actually does fix the issue.
So far so good the external voltage supply for the OpenVOX card has
arrived and I can
.
David Shauger
Vice President
678-317-9444 x5 - voice
404-886-7603 - cell
This email has been certified by Comodo
Email certification helps prevent identity theft
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.
-
It's all coming from the same carrier IP and the same SIP trunk. All are
set via static IP (No registrations). And nat is set to no (Everything is
on a public IP).
Has anyone else run across anything similar?
Thanks
David
--
[image: Ringfree Communications, Inc] http://ringfree.biz
Using the BLFs on Digium phones does not require the use of the Digium
Phone Module for Asterisk, or DPMA. SchmoozeCom (the FreePBX guys) use the
BLFs on Digium phones independently of the DPMA.
I am not sure why a previous response refers to this module as 'toxic'. It
is a free to use module
If the OpenVox card can supply the voltage, then it will a configuration
option (probably in hardware, like some jumpers) of the card itself.
I was going to point you to the Xorcom Astribank, which I know supplies the
required voltage.
All the best,
David
On 4 Aug 2014 13:16, Roberto Fichera
Please come back to let us know if this actually does fix the issue.
On 4 Aug 2014 14:36, Roberto Fichera ker...@tekno-soft.it wrote:
On 08/04/2014 01:21 PM, David Duffett wrote:
If the OpenVox card can supply the voltage, then it will a configuration
option (probably in hardware, like some
://www.asterisk.org/hello
asterisk-users mailing list
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*David Duffett*
Digium, Inc. · Director, Worldwide Asterisk Community
6 Landscape Close, Weston on the Green · Bicester
Are there any quality Outlook integrations for asterisk out there? The
closest I'm finding is at http://camrivox.com and they don't support
Outlook 2013.
dw
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This is defined in chan_sip.c. Simply edit the source file and recompile.
On Wed, Jul 23, 2014 at 1:39 PM, Nick Cameo sym...@gmail.com wrote:
Long story... Would be nice if we can remove this
on BYEs
X-Asterisk-HangupCause: Normal Clearing.
X-Asterisk-HangupCauseCode: 16.
Kind Regards,
.
On Wed, Jul 23, 2014 at 4:39 PM, David Lam software...@gmail.com wrote:
This is defined in chan_sip.c. Simply edit the source file and recompile.
On Wed, Jul 23, 2014 at 1:39 PM, Nick Cameo sym...@gmail.com wrote:
Long story... Would be nice if we can remove this
on BYEs
X-Asterisk
Reason header and use Reason header
if it is available.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Lam
*Sent:* Wednesday, July 23, 2014 5:07 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
We use standard sip ports all day long, and have had no issues with
employee phones on the verzion network.
On Jul 18, 2014 12:03 PM, Eric Wieling ewiel...@nyigc.com wrote:
Depends on the carrier. Verizon Wireless appears to activly block
SIP.G729 codec is needed on 3G and is a good idea
--
*David Pinedo García*
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_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users
.
This works from Asterisk 11.
On Fri, Jun 27, 2014 at 11:00 AM, David Pinedo dpin...@presenceco.com
wrote:
Hello,
Throwing calls from Asterisk to PSTN (via a VoIP gateway) some operators
sends an explaining audio, in situations as:
The phone number does is not assigned
The phone is powered
to implement my own
early media detector?
Thank you in advance
---
David Pinedo
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New to Asterisk? Join us for a live introductory webinar every Thurs
Thank you very much.
On 14 June 2014 00:33, Shaun Ruffell sruff...@digium.com wrote:
On Fri, Jun 13, 2014 at 12:54:14PM +1000, David Cunningham wrote:
Hello,
I'm getting the following errors when compiling dahdi-linux 2.6.2 under
Ubuntu 14.04 with kernel 3.13.0-24-generic.
I did
: *** [modules] Error 2
make: Leaving directory `/usr/src/dahdi-linux-2.6.2'
'make -C dahdi-linux-2.6.2 install' failed with 512.
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David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
, but I am not
seeing any AMI Hold event that corresponds to removing the call from Hold.
Is this the intended behavior, or am I missing something in how the call's hold
status should be tracked via the AMI?
Thanks!
David Huebner
Hi Rusty,
We found the problem - a configuration error. Thank you for the response.
On 29 May 2014 23:35, Rusty Newton rnew...@digium.com wrote:
On Thu, May 22, 2014 at 6:22 PM, David Cunningham
dcunning...@voisonics.com wrote:
Hello,
We have servers running Asterisk 1.8.20.1
[default]
Thanks for any advice.
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I've just deployed several VVX 600's with the Color Expansion Module.
And I'm having a minor issue with them.
Intermittently when a call comes into a ring group the user is
presented with the call pickup option associated with a BLF entry. Not
the normal answer/reject option.
I've explicitly
Built asterisk 11.8.1 on a Debian VPS. Testing using ekiga.
If h261 is checked in ekiga's video format list I have video, and
mouse over the video window shows it to be using h261.
But then I get the following lines a dozen or more times in the CLI:
[Mar 21 16:25:32]
On (21/03/14 13:28), Adrian Serafini wealwild...@wombit.com put forth the
proposition:
If h261 is checked in ekiga's video format list I have video, and
[Mar 21 16:25:32] WARNING[31818][C-0010]: file.c:1241
ast_writefile: No such format 'h261'
Ekiga can do SIP. Maybe try that? And
On (21/03/14 13:54), Steve Totaro stot...@totarotechnologies.com put forth
the proposition:
I found below here: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
If you have too many conferences, one CPU may not be able to mix all the
audio and you will have audio problems even if there
...@lists.digium.com] On Behalf Of David Woodfall
Sent: Friday, March 21, 2014 1:37 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ast_writefile: No such format 'h261', yet h261 is
the only video format that works.
On (21/03/14 13:28), Adrian Serafini wealwild...@wombit.com put forth
On (21/03/14 15:20), Adrian Serafini wealwild...@wombit.com put forth the
proposition:
On 03/21/2014 02:09 PM, David Woodfall wrote:
H.323 is a communications protocol like SIP. H261 is a codec like
ulaw or gsm. You do not need H323 unless you are using the H323
protocol INSTEAD of SIP
On (21/03/14 20:07), Dave Woodfall d...@dawoodfall.net put forth the
proposition:
On (21/03/14 15:20), Adrian Serafini wealwild...@wombit.com put forth the
proposition:
On 03/21/2014 02:09 PM, David Woodfall wrote:
H.323 is a communications protocol like SIP. H261 is a codec like
ulaw
On (21/03/14 15:20), Adrian Serafini wealwild...@wombit.com put forth the
proposition:
On 03/21/2014 02:09 PM, David Woodfall wrote:
H.323 is a communications protocol like SIP. H261 is a codec like
ulaw or gsm. You do not need H323 unless you are using the H323
protocol INSTEAD of SIP
Hello all.
I am currently using Asterisk 11.7.0 (also tried Asterisk 12, but same
behavior) and is having an issue when it comes to reINVITE on BYEs.
Apparently one of the SIP providers that I am using does not always process
reINVITEs correctly, and would return a 500 Internal Server Error
Is there another router in the mix? Put the cable modem in bridge mode and
attAch a real router.
http://401stblow.wordpress.com/2012/10/21/fixing-time-warner-cable-ubee-modem-connectivity-issues/
On Thursday, February 6, 2014, Mike Diehl mdiehlena...@gmail.com wrote:
I've got the registration
PM, Mike Diehl mdiehlena...@gmail.com wrote:
Unfortunately, we plug straight into the Ubee and the ISP will not support
any other modem.
GRRr..
Mike.
On Thu, Feb 6, 2014 at 12:34 PM, David Wessell da...@ringfree.biz wrote:
Is there another router in the mix? Put the cable
, Administrator TOOTAI ad...@tootai.net wrote:
Le 20/01/2014 03:51, David Cunningham a écrit :
Hi,
We have a Kamailio and Asterisk cluster, both machines being on a real
103.x IP address and also on a 172.x OpenVPN address.
The problem is that when Kamailo receives a call from the VPN and
forwards
/2014, at 7:18 pm, David Cunningham dcunning...@voisonics.com
wrote:
Hi Duncan,
Thank you for your reply. Here's the netstat:
[root]# netstat -udpln | grep asterisk
udp0 0 0.0.0.0:50000.0.0.0:*
6672/asterisk
udp0 0 0.0.0.0:4520
Hi Larry,
No, they are on separate machines.
On 21 January 2014 17:54, Larry Moore lmo...@omninet.net.au wrote:
Is Kamalio running on the same system as Asterisk?
On 21/01/2014 2:41 PM, David Cunningham wrote:
Hi Larry,
Thanks for the reply. We have all of those settings left out
Hi Andres,
Thanks for the idea. We did send bindaddr to the VPN address and restarted
Asterisk, but unfortunately that didn't solve the issue. Asterisk didn't
complain, but still the sip set debug on didn't show the packets.
On 22 January 2014 01:40, Andres and...@telesip.net wrote:
David
at 12:40 AM, David Cunningham
dcunning...@voisonics.com wrote:
Hi Paul,
Using ngrep/tcpdump shows the packet clearly going from the Kamailio
server
and arriving at the Asterisk server. This is why it's a mystery that
Asterisk doesn't see the call coming in. We tried removing
On 22 January 2014 09:11, Steve Edwards asterisk@sedwards.com wrote:
(Please don't top-post.)
On Wed, 22 Jan 2014, David Cunningham wrote:
We did send bindaddr to the VPN address and restarted Asterisk, but
unfortunately that didn't solve the issue. Asterisk didn't complain
list
To UNSUBSCRIBE or update options visit:
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Hi Paul,
Thanks for the reply. What are you looking for in the PCAP, that isn't in
the tcpdump earlier in the thread? I just want to make sure we gather the
information required.
--
David
Hi Paul,
The ngrep on the Asterisk server does show it being received. Have you any
idea what would prevent it getting from the network stack to Asterisk on
that machine?
On 21 January 2014 05:30, Paul Belanger paul.belan...@polybeacon.comwrote:
On Sun, Jan 19, 2014 at 9:51 PM, David
Hi Duncan,
The Asterisk machine also has a VPN IP address, so it has a route for 172.x
addresses to go to tun0 VPN interface.
On 21 January 2014 08:30, Duncan Turnbull dun...@e-simple.co.nz wrote:
On 21/01/2014, at 10:24 am, David Cunningham dcunning...@voisonics.com
wrote:
Hi Paul
are telling Asterisk to not allow the OS to pick the source IP
and hence the routing.
The *bindaddr options are seldom useful.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of David Cunningham
Sent: Monday
unfortunately.
On 21 January 2014 15:29, Paul Belanger paul.belan...@polybeacon.comwrote:
On Mon, Jan 20, 2014 at 4:24 PM, David Cunningham
dcunning...@voisonics.com wrote:
Hi Paul,
The ngrep on the Asterisk server does show it being received. Have you
any
idea what would prevent it getting from
, David Cunningham dcunning...@voisonics.com
wrote:
Hi Paul,
Using ngrep/tcpdump shows the packet clearly going from the Kamailio
server and arriving at the Asterisk server. This is why it's a mystery that
Asterisk doesn't see the call coming in. We tried removing the firewall (so
iptables -L
worth.
On 20/01/2014 10:51 AM, David Cunningham wrote:
Hi,
We have a Kamailio and Asterisk cluster, both machines being on a real
103.x IP address and also on a 172.x OpenVPN address.
The problem is that when Kamailo receives a call from the VPN and
forwards it to the Asterisk server
:
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is the Kamailio server's VPN address
103.y.y.y is the Asterisk server's real address
192.z.z.z is the calling phone's LAN address
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http://camrivox.com/products/flexor-cti-salesforce/
We've used this for a few clients.
On Fri, Jan 10, 2014 at 6:33 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi people
I'm just mailing to see what people are using for CTI solutions with
asterisk. Aslos, has anyone managed to integrate
No major issues. They're always very responsive. I'd get a demo from
them for the client and make sure that the feature set is a match. But
I always say that with 3rd party apps.
On Fri, Jan 10, 2014 at 10:39 AM, Tim Nelson tnel...@rockbochs.com wrote:
- Original Message -
for more details.
Looking that up, it says add to asterisk.conf
[options]
live_dangerously = yes
After doing this, and stopping and starting I
still get the message.
Whats up?
You want to avoid danger, so set live_dangerously = no.
Jerry
--
David M. Lee
Digium, Inc. | Software
I just checked my calendar, and - surprisingly - it's not April 1st!
On 4 Dec 2013 23:55, Gregory Malsack gmals...@coastalacq.com wrote:
I second that!
*Sent from my Verizon Wireless 4G LTE DROID*
Eric Wieling ewiel...@nyigc.com wrote:
Asterisk is Open Source, any company can port
I believe registration is in place, otherwise inbound calls would not work.
Also, registration is not required for outbound calls to work.
I would suggest cutting down your sip.conf profile to this minimal
configuration:
host=sipgate.co.uk
username=xxx
fromuser=xxx
insecure=invite,port
I would set a no-use flag in all extensions that you do not want to use
the h, and then test for it in the h extension itself - if it is set you
could just run the Hangup application.
On 28 Aug 2013 08:51, Grant Bagdasarian g...@cm.nl wrote:
Hello,
** **
We have a Kamailio SIP Proxy in
http://www.camrivox.com/products/flexor-cti-dynamics-crm/
--
Ringfree Communications
David Wessell
828-575-0030 x101
From: Steven Howes steve-li...@geekinter.netmailto:steve-li...@geekinter.net
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Hi All,
I'm hoping someone can recommend a method to integrate Microsoft CRM with
Asterisk. Preferably an open source product otherwise a commercial product.
Regards
David Klaverstyn
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_
-- Bandwidth and Colocation Provided
you're talking about.
I know it's not a lot of info, but hopefully you can turn up some
logging or packet captures to narrow down what's going on.
Thanks in advance,
Alex
--
David M. Lee
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out
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[image: Digium logo]
*David Duffett*
Digium, Inc
Does anyone have experience setting up an AudoCodes MP-X with an asterisk
(FreePBX based) system? I would be willing to pay a reasonable amount for
assistance with the MP-X device. I have remote access setup, so no one should
have to leave their comfy chair..
Thanks
David
I have never known them to not reply quickly. Email me offlist and I will give
you non generic email addresses.
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ringfree.biz
supp...@ringfree.biz
828-575-0030
On Jun 17, 2013, at 8:14 PM, Carlos Alvarez car...@televolve.com wrote:
We have licensed both products and sent a support request
You do not require DAHDI Linux or Tools if you do not have any TDM devices
unless you want to use MeetMe instead of ConfBridge.
Regards
David.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of luke devon
Sent: Saturday, 8 June
in the Community that have made
great contributions - if you have any nominations for this year, please let me
know.
REMEMBER: If you're serious about Asterisk, you'll be at AstriCon! Registration
is at http://www.asterisk.org/community/astricon-user-conference/register
All the best,
David
Digium
modem I would bridge it and put in a wrt54gl running
tomato. But that doesn't seem possible with this modem.
Does anyone have any experience with running this modem in conjunction with an
offsite asterisk server?
Thanks
David
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ringfree.biz
supp...@ringfree.biz
828-575-0030
On May 11, 2013
Quite a few SIP providers will have 911 testing functionality. Our main 911
provider lets you dial 933. Than they read back to you the address information
that is transmitted with the 911 call.
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Ringfree Communications
David Wessell
828-575-0030 x101
From: James Miller paramedi
Hi Matt,
You can't have multiple providers for inbound traffic. You can have multiple
providers for outbound traffic though.
Thanks
David
From: Matt Hamilton mistral9...@hotmail.commailto:mistral9...@hotmail.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Warren is correct. What I meant was you can't have multiple providers for the
same DID. You can get multiple DID's from different providers.
Thanks
David
From: Warren Selby wcse...@selbytech.commailto:wcse...@selbytech.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
of the created file are (mode ~umask).[1]
My guess is that the umask of your asterisk process is 022, which is very
typical. You'll have to play around with your umask settings and file
permissions to get things the way you want them.
[1]: http://linux.die.net/man/2/open
Ludovic BOUÉ
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David
Would love too hear more about this, as we are looking for a solution too.
Good comment.
Another feature suggestion
You might to ask the person to press 1 to confirm or 2 to leave a message if
the appointment is not going to be kept or 0 to reach the receptionist to
reschedule the
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