.
Brian
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to another SIP provider...
On Wed, Dec 23, 2009 at 12:55 PM, hadi motamedi motamed...@gmail.comwrote:
On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham
dcunning...@voisonics.com wrote:
Hadi,
You could use Asterisk as a sip server, it's installable on Windows.
Using sip set debug on might
CDR record!
== Spawn extension (tutorial, 4321, 1) exited non-zero on
'SIP/ivan-0a07dc80'
it says Failed to record Radius CDR record. Could you tell me ,
what's wrong with it?
2009/12/23 Olle E. Johansson o...@edvina.net:
23 dec 2009 kl. 11.25 skrev David Cunningham:
Shukun
, David Cunningham
dcunning...@voisonics.com wrote:
AsteriskWin32 does have SIP server functionality, same as the linux
version.
I can't think of any reason why having your CentOS Asterisk be both client
and server and register with itself wouldn't work.
Although I am wondering how much
/asterisk-users
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-15.
a=sendrecv.
a=ptime:20.
Any help would be much appreciated!
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Hello,
If you have canreinvite=no and a peer sends you a re-invite, what will
Asterisk reply with?
Thanks,
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have any idea why this is, or where I could go for more information?
Thanks for the help.
On Thu, May 13, 2010 at 11:06 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 05/13/2010 01:41 PM, David Cunningham wrote:
If you have canreinvite=no and a peer sends you a re-invite, what
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] WARNING[27482] channel.c: Exceptionally long voice
queue length queuing to Local/12126412...@asterisk-phone-7e3d;1
Thanks in advance!
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...@lists.digium.com] On Behalf Of David
Cunningham
Sent: Wednesday, May 19, 2010 3:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Cause and cure for Exceptionally long voice
queuelength queuing to Local?
Hello,
We're seeing lots of warnings like the following
I don't see anything in the SIP trace related to the warning messages.
Would anyone have any further tips?
Thanks for any help!
On Wed, May 19, 2010 at 9:12 PM, David Cunningham
dcunning...@voisonics.com wrote:
What should I expect see if it is the peer asking us to slow down RTP?
Thanks
Leif - thank you! Will try that.
On Fri, May 21, 2010 at 12:19 AM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
David Cunningham wrote:
Hello,
We're seeing lots of warnings like the following, running Asterisk
1.6.1.12. Does anyone know the cause or cure?
One explanation I've come
All,
Two questions:
1. Is there a limit on how long a call can be recorded for? For example is 4
hours a problem?
2. Can recording be stopped after a configured period of silence?
Thanks in advance,
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Danny, thank you!
On Wed, Sep 22, 2010 at 10:31 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Cunningham
*Sent:* Wednesday, September 22, 2010 4:28 PM
*To:* Asterisk Users
AGI program.
Thanks for any advice!
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Steve, that looks just the job, thank you very much.
On Wed, Dec 8, 2010 at 2:32 AM, Steve Edwards asterisk@sedwards.comwrote:
On Tue, 7 Dec 2010, David Cunningham wrote:
Is it possible to somehow 'bookmark' a place in a sound file? That is,
the user presses a key while a sound file
% increase) that would be great, rather than just
lots.
Also, are there any ATAs which are known to not work with progressinband =
yes? We have Polycom, Linksys and Audiocode.
Thanks for any advice,
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.
If you have any questions please don't hesitate to contact me directly.
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/drivers/dahdi/dahdi-base.o] Erreur
1
make[1]: *** [_module_/usr/src/dahdi-linux-2.4.0/drivers/dahdi] Erreur 2
make[1]: quittant le répertoire « /usr/src/linux-headers-2.6.34.6 »
make: *** [modules] Erreur 2
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Shaun,
CONFIG_MODULES wasn't enabled - thanks for the advice!
On Mon, Jan 31, 2011 at 4:02 PM, Shaun Ruffell sruff...@digium.com wrote:
On 1/30/11 8:45 PM, David Cunningham wrote:
I'm installing Asterisk with Dahdi on a server with a custom kernel
compile. I've got the kernel source
Hello,
We have a customer who upgraded from Asterisk 1.4 to 1.6.1.22 and is now
experiencing higher CPU utilization on their server. I can't see anything
wrong, so is this just expected with 1.6? Can anyone help explain it?
Thanks for any advice.
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http
volumes any better.
~Jared
On Wed, May 11, 2011 at 8:29 PM, David Cunningham
dcunning...@voisonics.com wrote:
Hello,
We have a customer who upgraded from Asterisk 1.4 to 1.6.1.22 and is now
experiencing higher CPU utilization on their server. I can't see anything
wrong, so is this just
-pickup-db70;2'
status is 'UNKNOWN'
The context doing the pickup looks like:
[product-pickup]
exten = _[0-9*#]!, 1, Pickup(${EXTEN}@product-phone)
Thanks for any advice,
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Hi all,
We have a project involving voice recognition, and will need a vocabulary of
10,000 words (actually names).
Can anyone recommend a product that works with Asterisk?
Thanks,
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Hi all,
I can't find the answer to this via google - is there some way to
permanently enable sip set debug on and agi set debug on in Asterisk? I
want this to be automatically enabled even after restarts.
Thanks for any advice.
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US toll-free
Kevin,
Thank you very much!
On 10 November 2011 00:15, Kevin P. Fleming kpflem...@digium.com wrote:
On 11/09/2011 04:22 AM, David Cunningham wrote:
Hi all,
I can't find the answer to this via google - is there some way to
permanently enable sip set debug on and agi set debug
.
Is this possible?
If not a confirmation that this is the case would be very helpful.
Thanks for any advice!
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can catch the signal
and do whatever you want to do.
Am 21.11.2011 07:38, schrieb David Cunningham:
Hello,
We would like to continue a Perl AGI after a Dial() it has done completes
following caller hangup. We would like to do this in the same AGI, and not
using a new AGI from the 'h
-info.org/wiki/view/Asterisk+cmd+Dial :
F(context^exten^pri): When the caller hangs up, transfer the called
party to the specified context and extension and continue execution.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 17:38 +1100, David Cunningham wrote:
Hello,
We would like to continue a Perl
to
the dialplan. I had incorrectly assumed you were doing the same.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 23:01 +1100, David Cunningham wrote:
Kingsley,
Thanks for the reply, but I am looking to continue within the same AGI
process and I believe that method would require starting
to persist
for the duration of the call?
Cheers,
Kingsley.
On Tue, 2011-11-22 at 14:27 +1100, David Cunningham wrote:
The strange thing is that we are using fast AGI, and for some reason
the AGI always exits when the caller hangs up - even when I set HUP to
IGNORE. If I set HUP
Hello,
Does SendFAX have the ability to put the caller ID and timestamp on the fax?
If so, is there a way to adjust the timezone used for the timestamp?
Thanks for any assistance.
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Steve,
Thanks for the reply.
Would anyone else know if Asterisk allows use of SpanDSP's time zone
conversion?
On 27 June 2012 00:24, Steve Underwood ste...@coppice.org wrote:
On 06/26/2012 10:24 AM, David Cunningham wrote:
Hello,
Does SendFAX have the ability to put the caller ID
format, that may do the trick.
Thank you for any advice.
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Hello,
We want the ability to choose from an AGI script whether or not to enable
faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone
suggest a workaround?
Thanks for any advice.
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:
On Thu, 3 Jan 2013, David Cunningham wrote:
We want the ability to choose from an AGI script whether or not to enable
faxdetect for calls over SIP or DAHDI.
What's the 'use case?'
You're going to call in and execute an AGI that will enable faxdetect for
future calls to this channel or other
...@lists.digium.com] *On Behalf Of *David Cunningham
*Sent:* Thursday, January 03, 2013 3:13 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] faxdetect on/off on the fly?
** **
Hello,
We want the ability to choose from an AGI script whether
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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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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
/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
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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
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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.
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, 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
[default]
Thanks for any advice.
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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
: *** [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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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
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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4227 (Map
https://goo.gl/maps/p25WF)
www.OntheNet.com.au http://www.onthenet.com.au/
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Cunningham
*Sent:* Tuesday, 18 August 2015 2:39 PM
*To:* Asterisk Users Mailing List
/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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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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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?
>&
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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t find any documentation to say what if anything is available. The
"aoc_enable" setting doesn't seem to have any effect in sip.conf.
Can anyone advise if there is any other support for AOC over SIP besides
Snom, and how to configure it?
Thank you,
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http://vo
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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with SIPSendCustomInfo but apparently it sends on all
active SIP channels, and is only available with TEST_FRAMEWORK.
Thanks for any advice.
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USA: +1 213 221 1092
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B, phone B answers the call, phone C dials
something to "steal" the call from B, and finally A and C are talking.
Searching on voip-info.org shows a "BristuffSteal" command but it's very
out of date (Asterisk 1.2).
Thanks in advance for any suggestions.
Kind regards,
--
David Cunn
d be able to use the Bridge dialplan application to do what you
> want.
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Bridge
>
> I use the CHANNELS function and the IMPORT function to find the channel to
> bridge to my caller.
>
>
> On Sun, Jul
.
Thanks for any advice.
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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Check out
to get 160
samples from write factory 0x1525f5d58ec8
[Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160
samples from write factory 0x1525f5d58ec8
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David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
. Thanks in advance for any help.
--
David Cunningham, Voisonics Limited
http://voisonics.com/
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New Zealand: +64 (0)28 2558 3782
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Check
Hi Steve,
What language is that please? We're using Perl and so far I haven't found
an equivalent there.
Thanks for your help.
On Fri, 7 Jun 2019 at 12:10, Steve Edwards
wrote:
> On Fri, 7 Jun 2019, David Cunningham wrote:
>
> > We have a need to record audio and allow the user
recording. Note
that only allowing # or * to end the recording won't work for us.
Does anyone know how we can tell which key ended the recording? Thanks in
advance for any help.
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28
Hi Steve,
Thank you very much for that information. The result is the key in ascii
perfectly!
On Fri, 7 Jun 2019 at 18:05, Steve Edwards
wrote:
> On Fri, 7 Jun 2019, David Cunningham wrote:
>
> > We're using Perl and so far I haven't found an equivalent there.
>
> On Thu,
RIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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11.25.3.
Thanks in advance for any assistance.
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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you in advance for any insight into this.
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out th
party puts the call on hold.
Thanks in advance for any assistance.
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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have items you fit as
well.
2. Provide your physical location, hours of availability, and indication of
hourly rate.
3. Let us know what other work you have during business hours.
Thank you,
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28
es the log files.
Does anyone know why?
Thank you in advance,
--
David Cunningham, Voisonics Limited
http://voisonics.com/
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New Zealand: +64 (0)28 2558 3782
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Hi Steve,
Thanks for the answer. Since that's what we already have configured, any
idea why it wouldn't work? As I said, when "asterisk -rx 'logger reload'"
is run it still rotates the log file.
On Wed, 20 May 2020 at 18:37, Steve Edwards
wrote:
> On Wed, 20 May 2020, David Cun
TE sent from 2.2.2.2:5060 to pstn.com
Does anyone know how this can be achieved?
Thanks in advance for your help,
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)
OK, thank you George.
On Sat, 24 Oct 2020 at 03:16, George Joseph wrote:
>
>
> On Thu, Oct 22, 2020 at 4:13 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hi George,
>>
>> Thank you for the response. I'm a little unclear on what you mean
that via the agi as well.
>
> On Wed, Dec 2, 2020 at 20:32 David Cunningham
> wrote:
>
>> Hi Dovid,
>>
>> We're using Enswitch so it uses AGI rather than a regular Asterisk
>> dialplan, but perhaps sending it to a custom-made Asterisk context with the
>&
> Does Asterisk send a 180 or a 183 with SDP? We have found that using these
> two lines help (where xc is a 500ms blank sound file)
> Exten => _X.,n, Progress()
> Exten => _X.,n, Playback(xc,noanswer)
>
>
> On Wed, Dec 2, 2020 at 4:30 PM David Cunningham
> wrote:
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