On Mon, 30 Aug 2010, J. Oquendo wrote:
Gordon Henderson wrote:
On Mon, 30 Aug 2010, J. Oquendo wrote:
I also posted a very effective iptables script some weeks ago if you care
to search the archives. It works and is extremely effective in blocking
these types of attacks - however, it will
On Tue, Aug 31, 2010 at 8:30 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
3) Contact the UPSTREAM of the attacking host?
Yes. No reply. And in the few times I've tried, I've only ever had a reply
from Amazon - some 18 hours after the flood started and then it took
another 12 hours
Hi everyone,
This is my first post to the list, although I am a long term user of Asterisk.
I have recently found a problem that I just can't seem to solve.
I have a client that has an Ubuntu x64 based Asterisk server with and ISDN
Dahdi interface and about 25 SIP handsets. Everything was
Hi!
After upgrading to Asterisk 1.6, we simply get no audio until the dialplan
finishes. On the Asterisk console, I can see that the sound file is indeed
playing, but we can't hear it. [...]
I have tried so many things that I have lost count, and I humbly ask the
collective intelligence of
Hi Alex,
I'm new to this list, but I had this problem too, and I solved it looking at
the codecs the sip handsets use, and then I converted the voice prompts to
that codec just like Philipp said..
Ondrej
On Tue, Aug 31, 2010 at 10:04 AM, Alex Ferrara a...@receptiveit.com.auwrote:
Hi
Hi all,
my asterisk will coredump in runing about ten days one time, and the
following is bt infor:
#0 0x00aac410 in __kernel_vsyscall ()
(gdb) bt
#0 0x00aac410 in __kernel_vsyscall ()
#1 0x00bead80 in raise () from /lib/libc.so.6
#2 0x00bec691 in abort () from /lib/libc.so.6
#3
On 8/20/10 1:24 PM, A J Stiles wrote:
On Wednesday 11 Aug 2010, Tilghman Lesher wrote:
On Wednesday 11 August 2010 03:59:28 A J Stiles wrote:
I'm having a problem with Asterisk 1.6.2.9 with the MySQL cdr addon.
With some calls, the value in the `billsec` field in the CDR is exceeding
the
Take a look here: http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue
http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue
Queue(queuename[|options][|URL][|announceoverride][|*timeout*][|AGI])
Hope it helps!
2010/8/30 Tino t...@sparksupport.com
Hello,
Is there any option to set the
On Tue, Aug 31, 2010 at 4:04 AM, Alex Ferrara a...@receptiveit.com.au wrote:
exten = 849,1,Playback(custom/ceh-meetingmsg)
exten = 849,n,Hangup
exten = 849,1,Progress()
exten = 849,n,Playback(custom/ceh-meetingmsg)
exten = 849,n,Hangup
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber:
On Tue, Aug 31, 2010 at 5:57 AM, jordan pan pylon...@gmail.com wrote:
my asterisk will coredump in runing about ten days one time, and the
following is bt infor:
Open an issue on https://issues.asterisk.org, besure to follow
doc/backtrace.txt and post all relevant information.
--
Paul
Here's the updated debug log.
http:/www.computerworkx.net/client/Document.txt
On 8/30/2010 2:55 PM, Paul Belanger wrote:
On Mon, Aug 30, 2010 at 2:48 PM, Todd Reesetrees...@gmail.com wrote:
Thanks for pointing out the misspelling. I've corrected that and still no
luck.
Create a new
On Tue, Aug 31, 2010 at 9:16 AM, Todd Reese trees...@gmail.com wrote:
Here's the updated debug log.
[Aug 30 15:27:41] NOTICE[11568] chan_sip.c: Call from '150' to
extension '6789542133' rejected because extension not found in context
'remote'.
So, again, a context problem. You can confirm by
asterisk*CLI dialplan show 6789542...@remote
There is no existence of 'remote' context
Command 'dialplan show 6789542...@remote' failed.
asterisk*CLI
On 8/31/2010 9:58 AM, Paul Belanger wrote:
dialplan show 6789542...@remote
--
From extensions.conf
[remote]
include = from-internal
include = dialout1
include = dialout2
include = dialout3
include = intercom
exten = 150,1,Macro(oneline,${EXTERNPHONE0})
[dialout1]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten =
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Reese
Subject: Re: [asterisk-users] help with dialplan
asterisk*CLI dialplan show 6789542...@remote
There is no existence of 'remote' context
Command
Todd--
There is probably some nifty anti-infinite-recursion code in the
extensions.conf parser,
to keep asterisk from going into infinite loops trying to descend into the
right context.
In your dialplan, [remote] includes dialout1, dialout2, dialout3, and each
of those
include remote.
Greetings all-
I have some dialplan code on an Asterisk 1.2.x box that basically dials a call,
then after call completion, runs a command via System(). However, I'm finding
that roughly 5% of the time, the System() command never executes and seems to
be on specific destinations.
I had already check on this. Thanks for the info, though.
On 8/31/2010 10:36 AM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Reese
Subject: Re: [asterisk-users] help with
On Tue, 31 Aug 2010, Tim Nelson wrote:
I have some dialplan code on an Asterisk 1.2.x box that basically dials
a call, then after call completion, runs a command via System().
However, I'm finding that roughly 5% of the time, the System() command
never executes and seems to be on specific
- Steve Edwards asterisk@sedwards.com wrote:
On Tue, 31 Aug 2010, Tim Nelson wrote:
I have some dialplan code on an Asterisk 1.2.x box that basically
dials
a call, then after call completion, runs a command via System().
However, I'm finding that roughly 5% of the time, the
Interesting things going on herel.
After your suggestions, Steve. I reran the dialplan show
16789542...@remote command with the below results.
Phone calls are geting the 404 error and the NOTICE on the console.
[Aug 31 11:06:46] NOTICE[11884]: chan_sip.c:20161 handle_request_invite:
Call
I am looking for pros and cons on the Intel Atom cpu. Has anybody been using
these in production? I am looking at an Atom D510 (dual core 1.6GHz, 1M cache)
to run maybe 25 to 30 extensions, 4 or 5 calls at once(maybe as high as 7
simultaneous calls), g729 all the way through except voicemail
On Tue, Aug 31, 2010 at 9:14 AM, Todd Reese trees...@gmail.com wrote:
Interesting things going on herel.
After your suggestions, Steve. I reran the dialplan show
16789542...@remote command with the below results.
Phone calls are geting the 404 error and the NOTICE on the console.
[Aug 31
Why not just copy the _1NXXNXX line into the remote context?
--
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On 31 Aug 2010, at 16:37, jmilli...@sentinelcommunications.com wrote:
I am looking at an Atom D510 (dual core 1.6GHz, 1M cache)
to run maybe 25 to 30 extensions, 4 or 5 calls at once(maybe as high as 7
simultaneous calls), g729 all the way through
Sounds fine to me. Reckon you could do that
On 31 Aug 2010, at 17:58, jmilli...@sentinelcommunications.com wrote:
On 31 Aug 2010, at 16:37, jmilli...@sentinelcommunications.com wrote:
I am looking at an Atom D510 (dual core 1.6GHz, 1M cache)
to run maybe 25 to 30 extensions, 4 or 5 calls at once(maybe as high as 7
simultaneous calls),
Just in case anyone is using Blockhosts
(http://www.aczoom.com/blockhosts/) with their Linux servers and
Asterisk here are the rules necessary to block invalid users:
asterisk-NoPeer:
r'Registration from .* failed for \'{HOST_IP}\' - No matching peer
found',
asterisk-NoAuth:
On Tue, 31 Aug 2010, jmilli...@sentinelcommunications.com wrote:
I am looking for pros and cons on the Intel Atom cpu. Has anybody been using
these in production? I am looking at an Atom D510 (dual core 1.6GHz, 1M
cache)
to run maybe 25 to 30 extensions, 4 or 5 calls at once(maybe as high
On Tue, 31 Aug 2010, Randy R wrote:
On Tue, Aug 31, 2010 at 8:30 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
3) Contact the UPSTREAM of the attacking host?
Yes. No reply. And in the few times I've tried, I've only ever had a reply
from Amazon - some 18 hours after the flood
Sounds fine to me. Reckon you could do that on a toaster ;)
S
Thanks, I needed to clean this keyboard anyway
--
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New to Asterisk? Join us for a live
On Tuesday 31 August 2010 07:49:19 Paul Belanger wrote:
On Tue, Aug 31, 2010 at 5:57 AM, jordan pan pylon...@gmail.com wrote:
my asterisk will coredump in runing about ten days one time, and the
following is bt infor:
Open an issue on https://issues.asterisk.org, besure to follow
On Tue, Aug 31, 2010 at 7:09 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
Their whole system is designed as a device to waste the time effort of
those trying to submit reports, etc. to them.
This is not the right list for the following comment, but vested
interests always ruin life.
Here, we have 2 aastra 9133i and on 9480i ct cordless. We use often the park
call feature of asterisk to transfer calls to one another.
But the 9480i ct cordless cannot pickup a parked call. When manually
entering 701 (parked call extention), the phone display Call failed (appel
écoué in
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nicolas Ross
Subject: [asterisk-users] Pickup parcked call from Aastra 9480i ct cordless
Here, we have 2 aastra 9133i and on 9480i ct cordless. We use often the
park
call feature of
Make sure your verbosity is set to at least 5 and try to see the CLI
output
on failure again. Are you sure the call is parked on 701 (not 702-720 as
defined in features.conf)?
Yes, after I can pick it up from my phone (9133i), and it works. I had
verbosity at 6 at the moment of testing.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nicolas Ross
Subject: Re: [asterisk-users] Pickup parcked call from Aastra
9480ictcordless
Make sure your verbosity is set to at least 5 and try to see the CLI
output
on failure again.
Yes, after I can pick it up from my phone (9133i), and it works. I had
verbosity at 6 at the moment of testing. When he enters 701, only his phones
displays Failed, nothing in asterisk. I can pickup after that on mine.
Is there a dial plan on the phone that you need to alter?
--
On 31 Aug 2010, at 18:10, Andrew Latham wrote:
Sounds fine to me. Reckon you could do that on a toaster ;)
Thanks, I needed to clean this keyboard anyway
Hehe. It's true though. I was amazed what our atom boards would do. We even
chucked transcoding/conferencing at them and they worked
Your (local phone) dialplan is not getting pushed out to the handset. Increase
the version number in your config to force it out to the handset...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of
Is the phone defined as a SIP extension/peer? If so, try sip set debug
peer xxx and try the call/pickup again.
Yes, and doing so, the phone could no longer dial out, bizare.
Yes, after I can pick it up from my phone (9133i), and it works. I had
verbosity at 6 at the moment of testing. When
On Tue, Aug 31, 2010 at 3:27 PM, Steve Howes steve-li...@geekinter.net wrote:
On 31 Aug 2010, at 18:10, Andrew Latham wrote:
Sounds fine to me. Reckon you could do that on a toaster ;)
Thanks, I needed to clean this keyboard anyway
Hehe. It's true though. I was amazed what our atom boards
On Tue, Aug 31, 2010 at 11:37 AM, jmilli...@sentinelcommunications.com wrote:
snip
simultaneous calls), g729 all the way through except voicemail will be wav
format for email purposes(requirement).
/snip
This will take up most of your CPU cycles, be sure to keep transcoding
to a minimum.
--
Hi Paul,
I tried adding Progress() to no avail. I still get no audio and below is what
comes up in the console.
-- Accepting call from '403xx' to '0812' on channel 0/10, span 1
-- Executing [0...@isdn-incoming:1] Dial(DAHDI/10-1, SIP/812,60) in
new stack
== Using SIP RTP CoS
You're probably not going to buy this, but if custom/ceh-meetingmsg is less
than 7 seconds long, it could be playing before the connection is
established.
--
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On Tue, Aug 31, 2010 at 5:50 PM, Alex Ferrara a...@receptiveit.com.au wrote:
Hi Paul,
I tried adding Progress() to no avail. I still get no audio and below is what
comes up in the console.
Try moving Progress() before the Dial(). If you Answer() the channel,
do you have the same problem?
Hello, i want to instal STUN Server in my Asterisk-PC. is it possible ? if
yes, how kann i do it ? where can i find STUN Program?
Thanks for your help.
--
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Hi folks,
My v1.6 Asterisk system logs all Call Detail Records to a PostgreSQL
database, including those handled by the Privacy Manager.
Unfortunately, even though I can use the CLI to see the information
being submitted by anonymous callers to satisfy the demands of the the
Privacy
On 1/09/10 11:27 AM, Jaap Winius wrote:
exten = jw,1,Verbose(-- CID is${CALLERID(num)})
exten = jw,n,GotoIf($[${CALLERID(num)}=anonymous]?true:false)
exten = jw,n(true),Set(CALLERID(num)=)
exten = jw,n(false),NoOp()
exten = jw,n,Verbose(-- CID is${CALLERID(num)})
exten =
On 25/08/10 7:35 PM, Julian Lyndon-Smith wrote:
Hey Matt, thanks for the response.
I know it sounds impossible. Hell, I sound like a user :) But it *is*
happening. And only on the cisco phones. We're trying to lab it up
right now. What should I be looking for in the sip debug ?
Just
On 25/08/10 8:54 PM, cov...@ccs.covici.com wrote:
Hi. I have a soft phone -- expresstalk-- on a computer in my network
and I use the internal ip address of the asterisk box to register the
phone. But using asterisk-1.8 between revisions 281912 and 281982 it
breaks -- after a few seconds of
On 28/08/10 10:18 AM, Russ Dill wrote:
My IAX2 trunk provider, Teliax, seems to be forcing early media. Early
media is cool and all, but my Asterisk install doesn't seem to be
fully supporting it. My initial setting was using Dial() to call all
of my dahdi (TDM400P) extensions. The results
On 30/08/10 2:48 PM, kawanobe tomohito wrote:
Hi
I want to know how to solve below an error case.
Uac cant's change username of from and digest header.
I tried to put a...@192.168.0.1 on username of sip.conf.but same error
returned.
You don't need to have the @192.168.0.1 in there -
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