lability.
4. A full CV is welcome.
Thank you,
--
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http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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t is 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:
7;s probably best if you 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&
Thank 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,
>>>
>>> D
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,
e media address
> 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
> learni
nk 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 or per peer?
Thanks for your help.
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 2
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
ackets 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,
>>
>> W
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 h
lability.
4. A full CV is welcome.
Thank you,
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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at
> 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" e
bug report.
Thanks in advance 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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on are you 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/${EXTEN}@sip-trunk-telekom)
>
> [image: image.png]
>
>
> On Thu, 21 Jul 2
tr,0,9});
> if ($var(prefix) == "force_tcp") {
> $rU = $(oU{s.substr,9,0});
> add_uri_param( "transport=tcp" );
> $fs = "tcp:" + $Ri + ":5060";
>
add_uri_param( "transport=tcp" );
> $fs = "tcp:" + $Ri + ":5060";
> }
> }
>
>
>
> On Wed, Jul 20, 2022 at 10:47 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hello,
>&g
host = 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
hat didn't seem to work. We are using chan_sip.
Thanks very much 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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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
C. Colp wrote:
> > 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
> > > 2 of
very much,
--
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 the new Ast
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 Cu
; -- Bandwidth and Colocation 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-us
AHDI 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
res_rtp_asterisk.c: Sent 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 Lim
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&g
On Wed, 27 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 re
d the call
somewhere different to all other calls.
Thanks very much,
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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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,
>>
&
s
t38_udptl = yes
t38pt_rtp = no
t38pt_tcp = no
Thanks 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 conne
sip.c: T38 state changed 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
. 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
de 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 tr
://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE 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 Zealan
know so we can 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
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
>
>
mplete-3.0.0+3.0.0/tools/xpp'
Makefile:1115: recipe for target 'all-recursive' failed
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory '/usr/src/dahdi-linux-complete-3.0.0+3.0.0/tools'
Makefile:664: recipe for target 'all' failed
make[1]: *** [all] Err
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 configur
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://voisonic
to do 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
>
> 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:
&g
ork around this issue?
Thank you in advance,
--
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http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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he 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 <
> dcu
Dovid 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,
>>
>> Doe
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
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
if it will work 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
INVITE 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: +6
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
7;" still rotates the log files.
Does anyone know why?
Thank you in advance,
--
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 Colocat
of the nice-to-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 Zea
e options in the Dial.
Thank 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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alled 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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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.
>
&g
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
d the 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:
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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_
-- B
t Called B can handle. We are using
Asterisk 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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P packets
themselves. Thanks in advance for any help.
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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ok.c: Failed 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
You should 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.
>
>
> O
lls phone 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,
lpful.
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 ou
can'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, Vois
me up with SIPSendCustomInfo but apparently it sends on all
active SIP channels, and is only available with TEST_FRAMEWORK.
Thanks for any advice.
--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
Australia: +61 (0) 2
Hi Jacek,
Thank you very much for the suggestion. Using SetVar and
CONNECTEDLINE(number) works.
On 12 December 2016 at 18:31, Jacek Konieczny wrote:
> On 2016-12-12 02:21, David Cunningham wrote:
>
>> Is there any equivalent of the CONNECTEDLINE function which can be
>&
Hello,
Is there any equivalent of the CONNECTEDLINE function which can be called
from an application using the AMI?
Thanks for any ideas.
--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
Australia: +61 (0) 2 8063 9019
iconhold.c:
Local/4171411@product-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
http://voisonics.com/
USA: +1 213 221
Shame, but thank you very much for the reply Joshua.
On 22 January 2016 at 10:26, Joshua Colp wrote:
> David Cunningham wrote:
>
>> Hello,
>>
>> Is it possible to mix PJSIP realtime and flat file configuration in
>> pjsip,conf?
>>
>> What we want
d 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
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
so
dwidth 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 mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com
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,
--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44
inal Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Ferrell
> Sent: Tuesday, 18 August 2015 4:38 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk 13 chan_sip
gt;
> This system is in semi-production, so there might be fluff in the log from
> other active calls.
>
>
>
> Brendan Ord
> OntheNet - Network Engineer
> P 07 5553 9222
> F 07 5593 3557
> Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map
> <https://goo
sity Parade Varsity Lakes Qld 4227 (Map
> <https://goo.gl/maps/p25WF>)
> www.OntheNet.com.au <http://www.onthenet.com.au/>
>
>
>
> --
> _____
> -- Bandwidth and Colocation Provided by http://www.a
l.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
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___
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>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
ting SRTP for audio, but they responded
without it!"
Thanks for any suggestions.
--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
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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,
--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
Thank you very much.
On 14 June 2014 00:33, Shaun Ruffell 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-
-2.6.2/drivers/dahdi] Error 2
make[1]: Leaving directory `/usr/src/linux-headers-3.13.0-24-generic'
make: *** [modules] Error 2
make: Leaving directory `/usr/src/dahdi-linux-2.6.2'
'make -C dahdi-linux-2.6.2 install' failed with 512.
--
David Cunningham
Hi Rusty,
We found the problem - a configuration error. Thank you for the response.
On 29 May 2014 23:35, Rusty Newton wrote:
> On Thu, May 22, 2014 at 6:22 PM, David Cunningham
> wrote:
> > Hello,
> >
> > We have servers running Asterisk 1.8.20.1 and 11.7.0, and on
il_zonemessages.conf
[default]
Thanks for any advice.
--
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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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2014 00:42, Administrator TOOTAI 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 Kamail
earlier in the thread? I just want to make sure we gather the
information required.
--
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http://voisonics.com/
USA: +1 213 221 1092
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On 22 January 2014 09:11, Steve Edwards 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
Hi Paul,
Thanks, we did try restarting Asterisk after the VPN was up but that didn't
solve the issue either.
On 22 January 2014 02:55, Paul Belanger wrote:
> On Tue, Jan 21, 2014 at 10:47 AM, Paul Belanger
> wrote:
> > On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham
&g
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Hi Larry,
No, they are on separate machines.
On 21 January 2014 17:54, Larry Moore 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
it came through a tunnel. Often it will be the tunnel
> interface address. Usually then we set the secondary address as the
> outbound proxy on the phone so the phone will also respond to it.
>
> Cheers Duncan
>
> On 21/01/2014, at 7:18 pm, David Cunningham
> wrote:
>
> Hi
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
5060
;received=203.z.z.z;rport=5060;branch=z9hG4bK274588850
From: ;tag=1880695235
To:
Call-ID: 1898224288
On 21 January 2014 16:56, Duncan Turnbull wrote:
>
> On 21/01/2014, at 6:40 pm, David Cunningham
> wrote:
>
> Hi Paul,
>
> Using ngrep/tcpdump shows the packet
idn't help unfortunately.
On 21 January 2014 15:29, Paul Belanger wrote:
> On Mon, Jan 20, 2014 at 4:24 PM, David Cunningham
> wrote:
> > Hi Paul,
> >
> > The ngrep on the Asterisk server does show it being received. Have you
> any
> > idea what would prevent it gettin
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
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 wrote:
> On 21/01/2014, at 10:24 am, David Cunningham
> wrote:
>
> Hi Paul,
>
> The ngrep on the Asteris
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 wrote:
> On Sun, Jan 19, 2014 at 9:51 PM, David Cunningham
> wrote:
mailio 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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David Cunningham, Voisonics
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