On 2017-04-18 05:21 PM, Duncan Turnbull wrote:
Sent from my iPhone
On 19/04/2017, at 11:43 AM, Ernie Dunbar <maill...@lightspeed.ca>
wrote:
On 2017-04-18 03:38 PM,
On 2017-04-18 03:38 PM, Duncan Turnbull wrote:
-- Original Message --
From: "Ernie Dunbar" <maill...@lightspeed.ca>
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'" <asterisk-use
te table.
Originalmeddelande
Från: Ernie Dunbar
Datum: 2017-04-19 00:25 (GMT+01:00)
Till: 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Rubrik: [asterisk-users] SIP connections over OpenVP
Hi everyone. I'm having some trouble with an OpenVPN tunnel that
isn't working *quite* as well as we'd hoped.
First, here's our technical details:
The OpenVPN server (v2.3.4-5+deb8u1) is a Debian 8 box behind a NAT
router. The router has UDP port 1194 forwarded
On 2016-07-14 16:40, Joshua Colp wrote:
Saint Michael wrote:
Many people are reporting the same issue, so it is not my
imagination.
Asterisk 13 above 13.1 is useless for anybody who relies on
res_odbc.so. As you know, after that version, the dropped the
complexity
of Pooling onto unix_odbc
On 2016-07-13 17:09, Ernie Dunbar wrote:
Hi everyone.
I'm trying to compile Asterisk with the smsq utility on Ubuntu 16.04
LTS, and while most things are compiling fine, smsq fails with the
following output:
root@test25:/usr/src/asterisk-certified-13.1-cert7/utils# make smsq
[CC] s
Hi everyone.
I'm trying to compile Asterisk with the smsq utility on Ubuntu 16.04
LTS, and while most things are compiling fine, smsq fails with the
following output:
root@test25:/usr/src/asterisk-certified-13.1-cert7/utils# make smsq
[CC] smsq.c -> smsq.o
[LD] smsq.o strcompat.o -> sms
Hi everyone.
It seems that all the documentation for Asterisk has become obsolete
when it comes to using the Monitor command on a call queue.
To the best of my knowledge, the way to get Asterisk to record a call
that goes into one of your call queues is by doing this in the dialplan:
exten
On 2016-02-17 16:28, Richard Mudgett wrote:
On Wed, Feb 17, 2016 at 5:56 PM, Ernie Dunbar
wrote:
On 2016-02-17 15:32, Richard Mudgett wrote:
On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar
wrote:
Hi everyone.
We have an Asterisk server running Debian Squeeze, with Asterisk
v1.8.13.1
On 2016-02-17 15:32, Richard Mudgett wrote:
On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar
wrote:
Hi everyone.
We have an Asterisk server running Debian Squeeze, with Asterisk
v1.8.13.1 (basically, the Debian Stable version for Squeeze, but
with some minor source code changes specific to our
Hi everyone.
We have an Asterisk server running Debian Squeeze, with Asterisk
v1.8.13.1 (basically, the Debian Stable version for Squeeze, but with
some minor source code changes specific to our site). We're trying to
upgrade to 11.13.1 (The Debian Stable version for Jessie), but I've run
int
com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie
Dunbar
Sent: Friday, November 20, 2015 3:25 PM
To: Asterisk Users
Subject: [asterisk-users] Which router/firewall would you use for a
virtual-PBX Asterisk installation?
Hi everyone.
We've got a fairly large base of customer
Hi everyone.
We've got a fairly large base of customers who use our Asterisk server
for phone service in a virtual PBX kind of way, where the server is
security hardened and exposed to the internet for them to connect to
remotely with SIP and IAX. It's certainly not the sort of affair where
w
This past holiday weekend has resulted in some real groaners when it
comes to bugs in our dialplan, making obvious the need for some changes
in our procedures.
First, our hours of operation for Christmas Eve, Christmas, Boxing Day
and New Year's Eve had changed with little to no notice. Okay,
Quoting Tim Nelson :
- Original Message -
Curiously enough, I can't do that at all on Voip3. Not span 3 of
course, because only span 1 should exist, but I can't execute "pri
show spans" either.
DING DING DING... we may have a winner. Do you have PRI support on
that box, meaning, d
Quoting Tony Mountifield :
In article <4feccd0c.1020...@fivecats.org>,
James Sharp wrote:
On 6/28/2012 3:53 PM, Ernie Dunbar wrote:
> We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a
> Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen),
>
Quoting Tony Mountifield :
In article <4feccd0c.1020...@fivecats.org>,
James Sharp wrote:
On 6/28/2012 3:53 PM, Ernie Dunbar wrote:
> We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a
> Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen),
>
Quoting Ioan Indreias :
On Thu, Jun 28, 2012 at 10:53 PM, Ernie Dunbar
wrote:
We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a
Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and
Voip3 has a Wildcard TE110P. Voip1 is the server that handles our
We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a
Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st
Gen), and Voip3 has a Wildcard TE110P. Voip1 is the server that
handles our PRI to the PSTN and we hope will allow us to failover to
other Asterisk servers
Hi List!
We have two Asterisk servers connected to a PRI, an old one and a new one.
The old server (voip1 let's call it) is running Asterisk 1.4.23, libpri
1.4.9, and DAHDI 2.1.0.4.
The new server (voip2) is running Asterisk 1.8.6, libpri 1.4.2 and DAHDI
2.4.0. We've had serious, show-stopping s
Quoting Carlos Chavez :
The disallow line must be set before any allow line.
Since Asterisk has no official G723 support you should not even be
trying to use that.
That's fantastic. I'll tell that to our SIP trunk provider right away.
Do you have the G.279 codec and license
This *should* be something that's easy to fix, but apparently I'm not
doing something right.
Our SIP long distance provider is telling us to only use formats G.723
and G.729, so I've set up their trunk configuration in sip.conf as such:
[t564]
type=friend
host=XXX.XX.56.4
context=default
di
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar
Sent: Tuesday, June 28, 2011 9:33 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Outgoing calls get dropped on high-latency
connections.
We're a VoIP provider essentially competing with our local incumbent
I think there is a bug in the Dial() application in Asterisk 1.6.2.17
that wasn't present in 1.4.23.1, and I'd like to see if anyone else
has this problem.
I've been able to reproduce this error: When you use the Dial()
command to send a call to both a SIP connection and a DAHDI
connectio
We're a VoIP provider essentially competing with our local incumbent
Telco, and a sizeable number of our customers use satellite internet.
As a result, these customers never have ping times less than 500ms,
and are often exceeding 2500ms.
I manually apply a "patch" to the Asterisk source co
I'm kind of at a loss to diagnose problems like this, yet we get them a lot.
- The ATA (Thomson 784 in this particular case) is logged into the
Asterisk server. 'sip show peer' shows their IP address, port, and
useragent.
- The ATA is connected directly to the internet (no NAT, but the sip
configu
> On 29/04/11 5:06 AM, Ira wrote:
>> At 05:56 AM 4/28/2011, you wrote:
>>> If I can install 1.8 and
>>> know that I can "turn off" things to get to 1.4 "solidness", then I
>>> don't
>>> have a problem with this kettle of fish. BTW, where does 1.10 fit into
>>> this
>>> conversation?
>>
>> Personall
see if that works also. its a tricky
> situation.
>
>
>
>
> On Wed, 2011-03-23 at 16:18 -0700, Ernie Dunbar wrote:
>> I have a Linksys 2102 ATA here that does call forwarding internally with
>> the *72 code, however our Asterisk 1.6.2.17 server doesn't forward the
I have a Linksys 2102 ATA here that does call forwarding internally with
the *72 code, however our Asterisk 1.6.2.17 server doesn't forward the
call properly. This is what shows up in the console when an incoming call
is made while the ATA is call-forwarded:
-- Called Username
-- Got SIP r
> On Mon, Mar 7, 2011 at 4:52 PM, Ernie Dunbar
> wrote:
>
>> [mya2billing]
>> dbhost = localhost
>> dbname = mya2billing
>> dbuser = a2billinguser
>> dbpass = REDACTED
>> dbport = 3306
>>
>
> Try adding "dbsock = /var/lib/mysql/mysql.
Okay, so here's the configuration I have for MySQL Realtime (Asterisk
version 1.6.2.17):
In /etc/asterisk/extconfig.conf:
sipusers => mysql,mya2billing,cc_sip_buddies
In /etc/asterisk/res_mysql.conf:
[mya2billing]
dbhost = localhost
dbname = mya2billing
dbuser = a2billinguser
dbpass = REDACTED
On our live server, running Asterisk 1.4.23.1 and DAHDI-Linux 2.1.0.4. On
occasion (not too rare, happens maybe once every month or two), the PRI
and/or DAHDI will stop working properly and we'll get repeated messages
like this:
[Feb 25 05:17:22] VERBOSE[9511] logger.c: -- B-channel 0/1 restar
>> On 16 February 2011 00:22, Ernie Dunbar wrote:
>>>> At 12:12 PM 2/15/2011, you wrote:
>>>>>I have two Aastra phones, a 6730 and a 6757, both connected to
>>>>> Asterisk
>>>>>v1.6.2.1. They can call each other's extensions (
> On 16 February 2011 00:22, Ernie Dunbar wrote:
>>> At 12:12 PM 2/15/2011, you wrote:
>>>>I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk
>>>>v1.6.2.1. They can call each other's extensions (and make and receive
>>>>
> At 12:12 PM 2/15/2011, you wrote:
>>I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk
>>v1.6.2.1. They can call each other's extensions (and make and receive
>>calls otherwise), but they cannot transfer calls, not even to outside
>
> I'm running 1.6.2.16.1 and have three Aas
> At 12:12 PM 2/15/2011, you wrote:
>>I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk
>>v1.6.2.1. They can call each other's extensions (and make and receive
>>calls otherwise), but they cannot transfer calls, not even to outside
>
> I'm running 1.6.2.16.1 and have three Aas
I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk
v1.6.2.1. They can call each other's extensions (and make and receive
calls otherwise), but they cannot transfer calls, not even to outside
extensions on the PSTN. The procedure I use is to accept a call on one
phone, press the
> On Wed, 9 Feb 2011, Ernie Dunbar wrote:
>
>>> We have a customer who wants to forward an extension to their cell
>>> phone, if and only if that extension is "unavailable", or when the
>>> Dial() command times out. However, should the Dial() command
htspeed7,10)
exten => s-BUSY,Voicemail(27)
exten => s-NOANSWER,Dial(DAHDI/g1/7788391675)
exten => s,n,Hangup()
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar
> Sent: Wedne
We have a customer who wants to forward an extension to their cell phone,
if and only if that extension is "unavailable", or when the Dial() command
times out. However, should the Dial() command return "busy" it should go
to voicemail instead.
As far as I know, the dialplan doesn't support this. C
This is a problem that is completely stumping me, and my understanding of
Asterisk dialplans tells me this should never be a problem. Moreover, this
scenario works on Asterisk 1.4 but not 1.6.
We have a customer with several Aastra 6731 phones. They want incoming
calls from the PSTN to work and th
We have an Asterisk server acting as a hosted PBX system for many clients,
and we're going through an upgrade to Asterisk 1.6 by moving our most
important (and complicated) clients one at a time.
But we're having a problem with one customer that I really can't explain.
I can place calls directly
> On Mon, Dec 20, 2010 at 5:39 PM, Ernie Dunbar
> wrote:
>> We have an issue with our Asterisk install where Asterisk produces many
>> Zombie processes (on the order of several hundred per minute) until
>> either
>> the Asterisk server is restarted (and the zombi
54 2104-7000
>
> ----- Mensagem original -
>
>
>
>> Am 20.12.2010 21:39, schrieb Ernie Dunbar:
>>> We have an issue with our Asterisk install where Asterisk produces many
>>> Zombie processes (on the order of several hundred per minute) until
>>>
> Am 20.12.2010 21:39, schrieb Ernie Dunbar:
>> We have an issue with our Asterisk install where Asterisk produces many
>> Zombie processes (on the order of several hundred per minute) until
>> either
>> the Asterisk server is restarted (and the zombies die a natural
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until either
the Asterisk server is restarted (and the zombies die a natural death), or
the kernel runs out of PID space (happens within hours) and brings the
system
I have some MP3 files that play well in any MP3 player I throw at them,
but when I try to make a MusicOnHold class with them, I get a continuous
stream of errors like this:
[Dec 2 13:20:31] WARNING[9120]: mp3/common.c:148 decode_header: Layer 2
not supported!
[Dec 2 13:20:31] WARNING[9120]: mp3/
Oh, this is most excellent. Although it means that my google-fu has failed
me. ;)
> On Fri, Nov 12, 2010 at 1:42 PM, Jonathan Thurman
> wrote:
> I didn't read the whole thing, but it looks pretty OK at a glance.
>
> http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html
>
> I hope that helps
We have two Asterisk servers. One is a live server supporting our
customers, and the other is a backup server that's being upgraded and
pressed into service. Both servers have a Digium TE405P T1 card in them,
and in order to test the T1 service on the backup server, I've created a
T1 crossover cabl
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