files without affecting the operation.
I realize that you can set astetcdir to another location in your custom
asterisk.conf but that means keeping copies of all the configs
elsewhere. I just want to keep copies of configs that I have modified.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
this gets set to true.
Interestingly a stable release of NetBSD does not have this issue
although it still has the second issue which I will start a separate
thread for.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
Voip: sip:da...@vex.net
This is the second issue I found while trying to install Asterisk on a
NetBSD box. I can't load the rtp module because HAVE_OPENSSL_SRTP
seems to be set. Is there some way to simply force this variab;e to be
unset from a configuration variable?
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
On Tue, 11 Jun 2013 18:42:07 +0300
Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Mon, Jun 10, 2013 at 04:06:27PM -0400, D'Arcy J.M. Cain wrote:
I am trying to build Asterisk on a NetBSD system but I am running
into two problems. The first only happens on an installation built
from
On Tue, 11 Jun 2013 18:43:55 +0300
Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Mon, Jun 10, 2013 at 04:10:23PM -0400, D'Arcy J.M. Cain wrote:
This is the second issue I found while trying to install Asterisk
on a NetBSD box. I can't load the rtp module because
HAVE_OPENSSL_SRTP seems
but I can't figure out where. Can anyone
suggest what area I should be looking?
Thanks.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth
on Koodo which uses the Telus network, the second largest,
and mine works fine.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth and Colocation
provider if you have this problem.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
there are not.
OP - can you clarify what actual command you are running?
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth and Colocation Provided by http
On Wed, 28 May 2014 12:19:10 +0200
Sander Smeenk ssme...@freshdot.net wrote:
Quoting D'Arcy J.M. Cain (da...@vex.net):
OP - can you clarify what actual command you are running?
I use 'core restart when convenient'.
Right. You said restart when convenient in your original email. I
tested
that they
are relevant but all of this was working under 11.10.2.
Thanks.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth and Colocation Provided by http
that interpretation. That's certainly what happened in 11.10.
I didn't see anything in the change logs that would suggest such a
drastic change in behaviour.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
On Thu, 7 Aug 2014 17:12:40 +0200
Asghar Mohammad asghar...@gmail.com wrote:
Your call is up on VoiceMail you should check dialstatus before
sending user to VoiceMail.
I removed the voicemail command from the dialplan and it was exactly
the same behaviour.
--
D'Arcy J.M. Cain
System
in the dark here but does core show channels show an inordinate
number of channels, especially channels that you know should be closed?
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
On Thu, 7 Aug 2014 10:12:02 -0400
D'Arcy J.M. Cain da...@vex.net wrote:
This just started after upgrading to 11.11.0. After a call is
completed (both ends hang up) the call still shows as active.
New data point - I just reverted to 11.10.2 without a single change to
the configuration
On Fri, 8 Aug 2014 11:01:50 +0200
Mikael Fredin mik...@wiraya.com wrote:
On 8 August 2014 04:50, D'Arcy J.M. Cain da...@vex.net wrote:
Shot in the dark here but does core show channels show an
inordinate number of channels, especially channels that you know
should be closed?
I get
. See the
thread Calls not hanging up. I reverted to 11.10.2 which fixed
my problem.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth and Colocation
because I
need the distribution files in the standard place for the above to work.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth and Colocation
the
distribution. Any idea where that template is meant to be used?
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth and Colocation Provided by http
overrule the compiled-in values.
Which, it seems to me, is exactly what one would want.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth and Colocation
the default?
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
trying to hack my switch?
Cheers.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
On Thu, 12 Feb 2015 16:39:55 +0200
Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Thu, Feb 12, 2015 at 09:25:39AM -0500, D'Arcy J.M. Cain wrote:
I know that it runs on other systems but do other ports get the same
attention? I have been running it on a NetBSD server for about a
year now
detection but
it still stops every few days or so. I have a cron job that tests for
it and restarts it when necessary.
Anyone else have experience on non-Linux systems?
Cheers.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
)
available for running unit tests. Are there already unit tests in the
distribution?
There is almost no Linux administration required once it is set up so
getting deep into the actual OS is not required.
Getting deep into an OS doesn't scare me.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
On Thu, 12 Feb 2015 16:39:55 +0200
Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Thu, Feb 12, 2015 at 09:25:39AM -0500, D'Arcy J.M. Cain wrote:
year now and while it mostly works it just crashes from time to
time with no explanation or core dump.
Use the option -g to get core dumps
the caller side unless registration is actually turned off
from the ATA and a sip unregister is issued.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth
device here and other clients are working behind a NAT gateway so I am
at a loss as to what might be wrong. Could it be the streaming?
Cheers.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
:
http://www.ipdeny.com/ipblocks/data/countries/all-zones.tar.gz
It has all of those blocks for all countries. I pick that up fresh
every week and block specific countries that I don't have clients in
but seem to be hitting me hard.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http
are just looking out for you but it only
takes one black hat.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth and Colocation Provided by http
RANT
We should be able to set up our systems based on the documentation and
not by let's try random things and see what works.
/RANT
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
()
CALLERID is a read only variable. Set the information in sip.conf in
the extension.
callerid=NAME 551212
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
format=wav49
fromstring=Vex.Net Voice Mail
nextaftercmd=yes
forcename=yes
pollmailboxes=yes
pollfreq=5
And for each extension I have this (sanitized):
1000 = 1234,D'Arcy,da...@example.com
That's the extension, PIN, name and email address.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http
is deprecated and that we should use CHANNEL instead. I
tried that and it said pbx.c: Function CHANNEL not registered. Does
that mean that this solution will eventually fail when SIPCHANINFO is
removed in some future release? I am running 11.17.1.
Cheers.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
tried setting a random string (xaccount) and reading it with
${ENV(xaccount)} but it's not an environment variable and didn't work.
Cheers.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
different ATAs, soft phones or SIP phones. Are my server
settings reasonable? Do I need to make specific requirements for the
client settings? Using a STUN server didn't seem to help. Is it a
good idea to specify it anyway?
Any help appreciated.
Cheers.
--
D'Arcy J.M. Cain
System Administrator
you can
try this? Specifically try to determine what RTP port number is being
negotiated when you have your zero-audio back from the remote party -
what RTP port and RTP server IP is he using at that moment on his
side?
I will check that.
Thanks for your suggestions.
--
D'Arcy J.M. Cain
System
likely since it didn't matter who started
the call.
I don't really care at this point. If 1% of the calls go through the
server when they didn't really need to it's no big deal.
Cheers.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
. Can anyone suggest a possible setup issue?
I have tried so many things but I am willing to try them again. Feel
free to make any suggestion no matter how silly. I really need to fix
this.
Cheers.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP
of setups.
Thanks for that. I was going nuts trying to figure this out.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth and Colocation Provided
-- SIP/thinktel-001d is making progress passing it to
SIP/416555-001b
0x7f7ff077d000 -- Probation passed - setting RTP source address to
206.80.250.102:26014
== Spawn extension (LocalSets, 416555, 6) exited non-zero on
'SIP/416555-001b'
--
D'Arcy J.M. Cain
System
On Mon, 24 Aug 2015 23:48:50 -0400
D'Arcy J.M. Cain da...@vex.net wrote:
exten = 200,1,Verbose(0,${CALLERID(all)} Calling PBX darcy)
same = n,GoTo(LocalSets,416555,1)
I tried changing the above to;
same = n,Dial(SIP/416555)
and
same = n,Dial(SIP/416555,,r)
Same problem
On Mon, 24 Aug 2015 23:48:50 -0400
D'Arcy J.M. Cain da...@vex.net wrote:
When I dial 416555 I get no ringback which I can sort of live with
since it gets answered pretty quickly but then when I dial 200 it
transfers me correctly to the 416555 extension but there is no
ringback
are having the same problem. See the recent thread on one way audio
started by me with my solution.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
it works fine but if I call it through the
virtual PBX it fails. I tried various combinations of Ringing and
'r' options and prematuremedia=no and progressinband=yes but
nothing seems to help. Can someone suggest a line of enquiry?
Cheers.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http
= 1400/500,0/15000
info = !950/330,!1400/330,!1800/330,0
stutter =
!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:d
> the webrtc client ignore the ringing when calling another SIP
> extension? Any ideas?
I had a similar problem. Turned out that my indications.conf file was
empty. Check that out.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da..
On Tue, 22 Sep 2015 11:32:14 -0300
Josué Conti <josueco...@gmail.com> wrote:
> Dear Joshua, my apologies, about this issue.
Excellent work. You managed to get your off-topic ad posted to the
list three times.
[text of ad deleted]
The lesson today, kids, is trim your responses.
--
D
oblems so I just remove
them all for good measure. Finally, I am not sure what the mechanism
is here but if it is like a goto then I think that you want the 's'
priority.
Or, I totally don't know what I am talking about and my education will
be advanced by the replies to this message. :-)
re is a delay between the last Rx and Tx suggesting that it thinks
that the numbers are being played but I don't hear them. Also, the
output to stderr does not appear in the logs.
Here is my environment:
- Asterisk 11.20.0
- NetBSD 7.0
- Python 3.4
Thanks in advance for any help or suggestions.
--
sort of proxy
software that will let me do that more efficiently? Is there some
magic DNS entries that can change it?
If it helps I am generating the configs from my client database for
asterisk as well as DNS so I don't care how complicated the configs get.
Thanks.
--
D'Arcy J.M. Cain
System
On Wed, 6 Jan 2016 23:21:44 -0500
"D'Arcy J.M. Cain" <da...@vex.net> wrote:
> Interestingly this led me down a different path. I added this to my
> script - comm('SET VARIABLE PYAST "hello world"') and displayed the
> varible in my dialplan and that worked
a number was just a
way to test the AGI. I didn't really need audio for my current
project. Now that I know that I can set variables I can continue.
However, I sure would like to know why SayWhatever isn't working. I
will start a new thread for that issue.
Thanks for the pointers.
--
D'Ar
On Thu, 7 Jan 2016 00:04:05 -0500
"D'Arcy J.M. Cain" <da...@vex.net> wrote:
> I am trying to figure out how to allow da...@example.com to be
> translated to dc2...@vex.net (out ISP domain) but I am at a loss to
I think I see where I can hook this.
same => n,V
nother SIP client Asterisk may
step out of the picture anyway not allowing you to record the call.
Turning that off forces all the calls to be proxied through you even if
they could talk directly.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@v
I to use the 'wait for
> digit' AGI command which allows the timeout to be specified in
> milliseconds.
If I understand it the OP has un-consumed input and is just looking for
the shortest possible time to read it. Would a read with a timeout of
zero do the job or would Asterisk optimize a
y calling it. Even then you don't know
whether an out of service is temporary or permanent.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth and
gt; configured to remove these prompts.
Define this local extension:
exten => *98,1,Verbose(0,${CHANNEL(peername)} calling voicemail)
same => n,VoicemailMain(${CHANNEL(peername)}@VoiceMail,s)
same => n,Hangup
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da..
swer message then they are in your mailbox. You
better have a password that they need to enter to continue.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth an
person's* unavailable message?
It's not for accessing another person's mailbox. It's for accessing
your own when you are away from home/office.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@v
u need to read the documentation a lot more. VoIP/SIP
is complicated. I certainly don't understand everything but I was able
to craft the above extension by reading up on extensions as well as
system variables.
--
D'Arcy J.M. Cain
System
On Thu, 4 Aug 2016 09:12:53 +0100
Nabeel <nabeelshik...@gmail.com> wrote:
> On 30 July 2016 at 19:32, D'Arcy J.M. Cain <da...@vex.net> wrote:
> > Not playing the prompt changes nothing. If someone presses '*'
> > while listening to your answer message then they ar
is entered after the 'mailbox' prompt.
>From outside phones. What happens when you dial "*98" from your own
phone.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
___
On Thu, 4 Aug 2016 14:03:39 +0100
Nabeel <nabeelshik...@gmail.com> wrote:
> I should add, a password is *always* asked if a password has been set.
> There isn't a way to bypass that.
Then something is wrong.
http://darcy.vex.net/star98.mp3
--
D'Arcy J.M. Cain
System Administrator,
er of times. Also, please
watch the attributions. I am not the one trying to make this work.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth
ten => *98,1,Verbose(0,${CHANNEL(peername)} calling voicemail)
same => n,VoicemailMain(${CHANNEL(peername)}@VoiceMail,s)
same => n,Hangup
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@
more
probability of being NAT related. Maybe it's a problem on the gateway
device. What is the modem/router?
P.S. The subject was driving me nuts. I had to correct it.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.ne
ones easily. Even if you
have to proxy the voice traffic (e.g. your phones are behind a NAT) it
should have no trouble with three concurrent calls.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.ne
bility-of-matching
which seems to imply that the above won't work and that all the calls
would go to the trunk. However, this is working as expected for me.
Did the behaviour change in the last four years or could I run into
problems with this setup? Perhaps I am misunderstanding the poster's
issu
00 extension and the generic one just happened to
have the same number of priorities. I have added explicit hangups.
Thanks.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:d
t matters I am running Asterisk 11.23.0 on NetBSD 7.0.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
On Sun, 4 Sep 2016 10:23:06 -0400
"D'Arcy J.M. Cain" <da...@vex.net> wrote:
> The docs for faxing seem a little light but I think I have managed to
> put something together. It mostly works but I have a couple of
> issues. The main one is identifying myself. Settin
On Thu, 1 Sep 2016 06:22:18 -0400
Mark Wiater <mark.wia...@greybeam.com> wrote:
> On 8/31/2016 9:57 PM, D'Arcy J.M. Cain wrote:
> > exten => 55,1,Verbose(Door buzzer calling)
> >> same => n,Set(toRing=)
> >> same => n,ExecIf($["${DEVICE_STA
ecIf"
Your application(s) is (are) not registered
Command 'core show application ExecIf' failed.
What module am I missing?
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
ation
same => n,Verbose(0,FAX charged to ${uid})
; for the log - actual billing uses the cdr table
same => n(faxfail),Hangup()
; all done
Please critique.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
5] WARNING[-1][C-0001fee7] app_dial.c: Unable to create
channel of type 'SIP' (cause 20 - Subscriber absent)
I am assuming that the voice mail is for the absent (unregistered) user.
> > exten => 55,1,Verbose(Door buzzer calling)
> >same => n,Dial(SIP/user1
;to remove the first &
>
> would do the work
That looks good and is easy to add and delete from the list. I will
give this a try one night this week. Not sure what that last line
would do if all of the phones are off but if they are the buzzer won't
be answered anyway.
the names of the
users. Other than that the above is the complete dialplan. See my
other message for details of those extensions. I didn't think that
there was anything relevant in that but I could be wrong.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:d
hoping
that there is an easier way so that I can create these types of
extensions for other clients easily as well as being able to add and
remove destinations quickly.
Cheers.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@v
ean by "Local channels?" They are local to the server.
Do you mean that I might be in the wrong context?
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
h it. It's hard to find
documentation for it other than the actual source code.
http://doxygen.asterisk.org/trunk/dc/d73/app__exec_8c-source.html
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
. Is there some way in the dialplan to set the
number that is used for that message? Is there a variable that can be
set before doing a GoTo to foo or foo3?
Thanks.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoI
hanks.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Join the Asterisk Community at
84 matches
Mail list logo
