A full CV is welcome.
Thank you,
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New Zealand: +64 (0)28 2558 3782
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Check o
happening.
>
> On Mon, Apr 17, 2023 at 8:52 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hi Joshua,
>>
>> Thank you for that. From the code it kind of looks like
>> STRICT_RTP_LEARN_TIMEOUT is a minimum, not a maximum:
read the logic[1]. There's an entire comment
> that talks about how it works.
>
> [1]
> https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158
>
> On Mon, Apr 17, 2023 at 7:10 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> H
you!
On Thu, 2 Mar 2023 at 12:32, Joshua C. Colp wrote:
> On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp wrote:
>
>> On Tue, Feb 28, 2023 at 9:50 AM David Cunningham <
>> dcunning...@voisonics.com> wrote:
>>
>>> Hello,
>>>
>>> Does a
Thank you Joshua.
On Thu, 2 Mar 2023 at 12:32, Joshua C. Colp wrote:
> On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp wrote:
>
>> On Tue, Feb 28, 2023 at 9:50 AM David Cunningham <
>> dcunning...@voisonics.com> wrote:
>>
>>> Hello,
>>>
>&g
Hi Joshua,
Thanks for that. The naming is a little confusing as "no'' makes it sound
like it's "not strict" - good to know though. Is it possible to set
strictrtp to no for just one peer?
On Wed, 1 Mar 2023 at 02:57, Joshua C. Colp wrote:
> On Tue, Feb 28, 2023 at 9:50 A
> of Asterisk A because it's getting RTP from it.
>
> Note we have "canreinvite = no" in sip.conf, but I don't think that's
> relevant to the problem.
>
> Can anyone suggest how to prevent this problem? Is it possible to turn off
> learning the media address per call
o the problem.
Can anyone suggest how to prevent this problem? Is it possible to turn off
learning the media address per call or per peer?
Thanks for your help.
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand:
Okay, thanks very much for your help Joshua.
On Mon, 31 Oct 2022 at 10:07, Joshua C. Colp wrote:
> On Sun, Oct 30, 2022 at 5:00 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hi Joshua,
>>
>> Thanks very much. I presume this is the relevant
ckets from the Asterisk log. Can anyone see an
issue that would cause the error?
Thanks in advance.
On Sat, 29 Oct 2022 at 12:03, Joshua C. Colp wrote:
> On Fri, Oct 28, 2022 at 6:28 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hello,
>>
>> We
error is that call 1
was set up with "CSeq: 954698786 INVITE", whereas the re-INVITE Asterisk
sends with the P-Asserted-Identity has "CSeq: 102 INVITE". Why is Asterisk
resetting the CSeq on the re-INVITE, and doesn't this appear to be
incorrect?
Thanks in advance for any help.
--
A full CV is welcome.
Thank you,
--
David Cunningham, Voisonics Limited
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USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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Check o
t; asterisk a restart... :S
>
> On Wed, Jul 27, 2022 at 6:21 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hello,
>>
>> We have an Asterisk 13.38.2 server which today started giving "we
>> couldn't allocate a port for RTP" errors.
.
Thanks in advance for any advice.
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David Cunningham, Voisonics Limited
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Check
using ?
> please show: asterisk -rx "sip show peer sip-peer"
>
> I checked...
> I use UDP and TCP, my phone via UDP, telekom via TCP and works
>
>
> same => n,dial(SIP/${EXTEN}@sip-trunk-telekom)
>
> [image: image.png]
>
>
> On Thu, 21 Jul 2022 at 23:
if ($var(prefix) == "force_tcp") {
> $rU = $(oU{s.substr,9,0});
> add_uri_param( "transport=tcp" );
> $fs = "tcp:" + $Ri + ":5060";
> }
>
_param( "transport=tcp" );
> $fs = "tcp:" + $Ri + ":5060";
> }
> }
>
>
>
> On Wed, Jul 20, 2022 at 10:47 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hello,
>>
>>
= final.destination.com
transport = tcp
outboundproxy = our.proxy.com
On Fri, 22 Jul 2022 at 01:23, Henning Follmann
wrote:
> On Thu, Jul 21, 2022 at 02:46:07PM +1200, David Cunningham wrote:
> > Hello,
> >
> > We have an Asterisk dial which sends the call via a proxy usin
hat didn't seem to work. We are using chan_sip.
Thanks very much for any advice.
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David Cunningham, Voisonics Limited
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Thank you Thomas.
On Mon, 18 Jul 2022 at 12:24, Thomas Ray wrote:
> Moving to chan_pjsip solves this problem.
>
>
>
> *From: *asterisk-users on
> behalf of David Cunningham
> *Reply-To: *Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-user
ote:
> > On Fri, Jul 15, 2022 at 1:37 AM David Cunningham <
> dcunning...@voisonics.com>
> > wrote:
> >
> > > Hello,
> > >
> > > We have an Asterisk server with 3 IP addresses, and need to listen on
> only
> > >
much,
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David Cunningham, Voisonics Limited
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New Zealand: +64 (0)28 2558 3782
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Check out the new Asterisk
Hi Joshua,
You're right, it was a firewall problem. One of those things where testing
a change in one place throws up a previously unseen problem somewhere else!
Thanks for the tip.
On Thu, 19 May 2022 at 21:18, Joshua C. Colp wrote:
> On Thu, May 19, 2022 at 6:04 AM David Cunning
olocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
>
DI running on the server:
# asterisk -rx 'dahdi show version'
DAHDI Version: 3.0.0 Echo Canceller:
# asterisk -rx 'dahdi show status'
Description Alarms IRQbpviol CRCFra
Codi Options LBO
On Thu, 19 May 2022 at 15:51, David Cunningham
wrote:
> Hello,
>
nt RTP
packet to bb.bb.bb.20:35412 (type 00, seq 020934, ts 105920, len 000160)
[May 17 20:26:24] VERBOSE[17725][C-0027] res_rtp_asterisk.c: Got RTP
packet from bb.bb.bb.20:35412 (type 00, seq 029996, ts 102760, len 000160)
Thanks very much,
--
David Cunningham, Voisonics Limited
http://voisonics.com
Hi Joshua,
Thanks for the reply. In this case we get a special SIP header in the 302,
but I guess we'll need to find another solution to use it.
On Wed, 27 Apr 2022 at 21:27, Joshua C. Colp wrote:
> On Wed, Apr 27, 2022 at 5:33 AM David Cunningham <
> dcunning...@voisonics.com> wr
Apr 2022 at 18:57, Jon Bonilla (Manwe)
wrote:
> El Wed, 27 Apr 2022 12:27:03 +1200
> David Cunningham escribió:
>
> > Hello,
> >
> > Does anyone know of a way to have a call go to a particular context when
> a
> > 302 Moved is received in response to
the call
somewhere different to all other calls.
Thanks very much,
--
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Hi Joshua,
Thank you for that. In the end it seems to have been a firewall blocking
the UDPTL ports.
On Thu, 24 Mar 2022 at 11:15, Joshua C. Colp wrote:
> On Wed, Mar 23, 2022 at 7:07 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hi Joshua,
&
again.
On Fri, 18 Mar 2022 at 21:59, Joshua C. Colp wrote:
> On Fri, Mar 18, 2022 at 12:27 AM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hello,
>>
>> We have a problem where one fax ATA connected to Asterisk works, and
>> another ATA wi
nged to
3 on channel SIP/xx.xx.246.70:5060-0030e0a1
Notice the difference in the "T38 state changed to" values. Does anyone
know what a value of 1, 2, or 3 means? I tried to find out from the
Asterisk source code but it wasn't obvious.
Thank you in advance for any tips.
--
David Cunn
. I build
> from git and it works every time. I will try to look at my scripts and post
> later exactly what I do.
>
> On Mon, Feb 21, 2022 at 20:52 David Cunningham
> wrote:
>
>> Hello,
>>
>> I see some emails about a Dahdi compilation problem with
>> "li
Hello,
I see some emails about a Dahdi compilation problem with "linux/pci-aspm.h:
No such file or directory" two years ago, which suggest trying the "next"
branch.
Did this change go into a Dahdi release, and if so which version number(s)
please?
Thank you,
--
David C
to add the features we
need is what we're looking to hire someone for.
Thanks.
On Fri, 12 Nov 2021 at 13:20, David Cunningham
wrote:
> Hi Antony,
>
> Thanks for the suggestion. I didn't get a response on my request to join
> the asterisk-dev mailing list. I'll try asterisk
k.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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discuss
pricing and your Asterisk development experience.
If anyone has ideas for other places to advertise this request let me know!
Thanks very much,
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
//www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> ast
ols'
Makefile:664: recipe for target 'all' failed
make[1]: *** [all] Error 2
make[1]: Leaving directory '/usr/src/dahdi-linux-complete-3.0.0+3.0.0/tools'
Makefile:9: recipe for target 'all' failed
make: *** [all] Error 2
--
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USA: +1 213 221 1
Hi Steve,
Thanks for that. Perhaps the change to res_fax might help us? I'm hoping
someone can say whether or not for sure.
On Wed, 30 Dec 2020 at 11:00, Steve Edwards
wrote:
> On Wed, 30 Dec 2020, David Cunningham wrote:
>
> > Would anyone be able to tell us how to configure
this option for calls
arriving via chan_sip? Is it just a matter of setting the FAXOPT(faxdetect)
variable in the dialplan? What we'd like to do is restrict fax detection to
the first N seconds of a call.
Thanks very much for any advice,
--
David Cunningham, Voisonics Limited
http://voisonics.com
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:
this issue?
Thank you in advance,
--
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OS to handle
> it for you? So long as asterisk isn’t calling bind() (or is calling with
> 0.0.0.0) I would imagine adding a route for the peer, with your normal
> gateway, and the correct device would work.
>
> On Thu, Oct 29, 2020 at 10:04 PM David Cunningham <
> dcunning
Bender wrote:
> Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass
> it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio
>
> On Thu, Oct 29, 2020 at 20:44 David Cunningham
> wrote:
>
>> Hello,
>>
>> Does any
. There is the
"extenip" setting, but it's really designed for NAT, and can only appear in
the [general] section.
Any suggestions would be greatly appreciated.
On Sat, 24 Oct 2020 at 09:43, David Cunningham
wrote:
> OK, thank you George.
>
>
> On Sat, 24 Oct 2020 at 03:16
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
in a device that could be the answer.
On Fri, 23 Oct 2020 at 00:13, George Joseph wrote:
>
>
> On Wed, Oct 21, 2020 at 9:16 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hello,
>>
>> We have an Asterisk server with two public IP addr
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)
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
es the log files.
Does anyone know why?
Thank you in advance,
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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
you in advance for any insight into this.
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Check out th
party puts the call on hold.
Thanks in advance for any assistance.
--
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New Zealand: +64 (0)28 2558 3782
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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,
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
RIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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11.25.3.
Thanks in advance for any assistance.
--
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. Thanks in advance for any help.
--
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Check
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
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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
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
.
Thanks for any advice.
--
David Cunningham, Voisonics Limited
http://voisonics.com/
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New Zealand: +64 (0)28 2558 3782
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Check out
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,
--
David Cunningham, Voisonics
http://vo
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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http://voisonics.com/
USA: +1 213 221 1092
Australia: +61 (0) 2 8063 9019
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/
USA: +1 213 221 1092
Australia: +61 (0) 2 8063 9019
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
--
David Cunningham, Voisonics
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USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia:
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.
--
David Cunningham, Voisonics
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USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
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?
>&
for a live introductory webinar every Thurs:
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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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USA: +1 213 221 1092
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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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asterisk-users mailing list
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by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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webinar every Thurs:
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asterisk-users mailing list
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for any suggestions.
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New
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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USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
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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
: *** [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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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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, 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
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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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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
:
http://lists.digium.com/mailman/listinfo/asterisk-users
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USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
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-- Bandwidth and Colocation
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://voisonics.com/
USA: +1 213 221 1092
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