Nope,
Its a totally normal self-built Asterisk.
Dan
Zeeshan Zakaria zisha...@gmail.com wrote:
Do you use FreePBX by any chance?
Zeeshan A Zakaria
--
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On 2010-10-16 6:38 PM, Dan Journo
I think I've seen this where I am trying to start another instance of
asterisk using safe_asterisk, when I already have an instance running
Julian
On 16 October 2010 22:36, Dan Journo d...@keshercommunications.com wrote:
Hi,
Does anyone know where this is suddenly coming from?
--
mù:l;:kj,nb hgyuè
2010/10/16 Frank Tarczynski ft...@mindspring.com
I'm running an AsteriskNow V1.7.1 with both a PSTN connection and fax
machine. Both are connected to a DAHDI board. I'd like to route
incoming PSTN fax calls to the extension of the fax machine and process
non-fax calls
Some service is definitely connecting to your asterisk using AMI. Such
services use username/password described in manager.conf. Usually its is
some monitoring service. Although the message says 'remote UNIX connection'
but it can be very well something from localhost. I would suggest to use
- Original Message -
When we designed our systems on asterisk we designed it to me multi-tenant. Se
we use customer prefixes on all extensions. This allows us to have multiple
customers using the same extension pools. It also reduces the hack foot print
as hackers must know the
Some service is definitely connecting to your asterisk using AMI. Such
services use username/password described in manager.conf. Usually its is some
monitoring service. Although the message says 'remote UNIX connection' but it
can be very well something from localhost. I would suggest to
Hi,
I can use TDMoE for trunking of Asterisks. it is no problem.
In use of TDMoE for channel bank, my problem is:
All of channels in trunk are OFFHOOK!
Can you help me?
On 10/14/10, Luis Antonio Prata Barbosa luispratalis...@gmail.com wrote:
Hi Karim,
Here you find
Do you have freepbx anywhere it always tries to connect -- via a socket
I think and it usually uses the manager, so if you disable the manager
it will break things. Also take the port stanza off of the tcpdump and
you will soon see what is connecting. You will get other stuff, but
this will tell
Dear friend,
Welcome to our website (www.ebay365.org), We're in the wholesale and
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you. Shopping at our website more favorable, high quality products,
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On Sun, Oct 17, 2010 at 6:55 AM, cov...@ccs.covici.com wrote:
Do you have freepbx anywhere it always tries to connect -- via a socket
I think and it usually uses the manager, so if you disable the manager
it will break things. Also take the port stanza off of the tcpdump and
you will soon
- Original Message -
Sounds like either FreePBX or some other script using astmanproxy or
just the AMI in general. Another possible cause is a script (or
terminal) constantly accessing asterisk -r or rasterisk (+ any
other arguments) to either run an Asterisk CLI command, or to just
SSH??
JN
Dan Journo wrote:
Hi,
Does anyone know where this is suddenly coming from?
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX
I'm really struggling with this DTMF issue.
In order to test it, I've tried a few different providers and DTMF RFC2833 does
work with any of them, even though a few of them insist that it is.
Is this a bug with 1.4.36?
Has anyone else experienced this problem?
The Asterisk CLI is showing the
On Sun, Oct 17, 2010 at 3:59 AM, Olivier oza_4...@yahoo.fr wrote:
mù:l;:kj,nb hgyuè
2010/10/16 Frank Tarczynski ft...@mindspring.com
I'm running an AsteriskNow V1.7.1 with both a PSTN connection and fax
machine. Both are connected to a DAHDI board. I'd like to route
incoming PSTN fax
Whats payload used for in rfc2833?
I'm wondering if that is incompatible.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: 17 October 2010 14:54
To: Asterisk Users Mailing List -
On Sun, 2010-10-17 at 09:53 -0400, Dan Journo wrote:
I'm really struggling with this DTMF issue.
In order to test it, I've tried a few different providers and DTMF RFC2833
does work with any of them, even though a few of them insist that it is.
Is this a bug with 1.4.36?
Has anyone else
Have you tried relaxdtmf or rfcc2833compensate?
Just tried it, but it didnt make a difference.
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Well, I connected my sip phone directly to the provider and totally skipped the
asterisk server.
DTMF rfc2833 worked fine!
Looks like Asterisk is doing something that's preventing it from working.
However, looking at the tcpdump from asterisk, it all looks fine.
Any ideas?
Thanks
Dan
I took a look in the source -- it is definitely asterisk -r (or
rasterisk) and not AMI. AMI logs something like Manager 'username'
logged on from 127.0.0.1.
Check the timing between calls and see if a pattern appears. If so,
it is some sort of cron/scheduled job. If not, keep looking!
-M
--
Hello,
I'm trying to conect two 1.6.2.13 Asterisk server with IAX.
This is my configuration:
Asterisk A:
iax.conf
register = coiax:pa...@69.164.207.166
[smiax]
type=friend
host=dynamic
trunk=yes
secret=pass2
context=phones
deny=0.0.0.0/0.0.0.0
permit=69.164.207.166/255.255.255.255
On Sat, 2010-10-16 at 17:36 -0400, Paul Belanger wrote:
On Sat, Oct 16, 2010 at 4:59 PM, Frank Tarczynski ft...@mindspring.com
wrote:
Any pointers to share?
chan_dahdi.conf
faxdetect=incoming
extensions.conf
exten = fax,1,Dial(DAHDI/4)
This is what I do and it works great, as long
Karim,
Try these 2 configurations for FXS and FXO hardware:
For FXS hardware
Remote asterisk (channel bank):
Configure FXS to immediate answer.
use s,1,Dial( dahdi / TDMoE_channel# ) in FXS context
use s,1,Dial( dahdi / FXS_channel#) in TDMoE_channel# context
Configure TDMoE_channel# to use
Dan,
I have to say, with 1.6.2.13 I am having the same or similar problems.
I switched the affected customers to inband, haven't had time
to delve too deep into the problem.
what I did see though, is that rfc2833 are indeed being sent, but
not recognized by two of the sip providers we use.
Ron
Hi ,
Is it possible to have two meetme room in asterisk 1.6 which each one have a
different language? I mean, one room the annoucement is in Portuguese an
another in english?
Today I can change over the sip.conf and it is valid for all room.
regards!
Att,
Flavio Roberto Miranda
Hi Flavio,
try with this funtion before the line with the english meetme application
Set(CHANNEL(language)=en)
and
Set(CHANNEL(language)=pr)
before the line with the portugues meetme application
Regards
- Bakko--
_
--
hi Bakko,
thanks!
Acctualy, I had tried this but still don´t works!
[conference]exten = 1001,3,MeetMe(1001,ipdM)exten =
1001,4,Set(CHANNEL(language)=pt_BR)exten =
1001,5,Playback(pt_BR/vm-goodbye)exten = 1001,6,Hangup
this is my config!
What´s wrong?
thanks again!
Att,
Flavio Roberto
Hi Flavio
is:
[conference]
exten = 1001,3,Set(CHANNEL(language)=pt_BR)
exten = 1001,4,MeetMe(1001,ipdM)
exten = 1001,5,Playback(vm-goodbye)
exten = 1001,6,Hangup
Regards
- Bakko--
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It works!!!
Thanks a lot!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
From: asannu...@gmail.com
To: asterisk-users@lists.digium.com
Date: Sun, 17 Oct 2010 17:56:37 -0500
Subject: Re: [asterisk-users] Meetme
Hi Flavio
is:
[conference]
On Sun, Oct 17, 2010 at 4:16 PM, Ron Arts ron.a...@neonova.nl wrote:
I have to say, with 1.6.2.13 I am having the same or similar problems.
I switched the affected customers to inband, haven't had time
to delve too deep into the problem.
I recommend reviewing [1] and look for possible
I'm having trouble getting an IAX2 connection between a couple of servers. Ican make calls from server B to server A, but when I call from Server A to serverB, I get "No authority found".On ServerA I am running Asterisk 1.6.2.9On ServerB I'm running 1.6.2.13Any hints for me? The registrations in
Just thought I'd let everyone know I've got a new beta version of app_swift up
for Asterisk 1.8 on http://forge.asterisk.org.
- Darren
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