have no idea
who can help.
Nobody, if you don't post them somewhere.
Steve
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On Fri, 14 Jan 2011, Tom Rymes wrote:
While we're at it, can someone please tell me whether I should be using vi or
emacs? ;-)
What 'rymes' with flame bait?
--
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Steve Edwards sedwa
??
The queue_log table contains exactly that information - Along with a
few other events, it indicates when a caller joined a queue, and when
an agent gets given the call. Take the difference between the 2 times
and you have the number that you need.
Cheers,
Steve
,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
on hand, but 1.2 1.6 use set().
Also, just a suggestion to make your dialplan more maintainable, check out
the 'n' priority instead of explicitly numbered priorities.
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Steve Edwards sedwa
the priority on the 'macro' line.
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Newline Fax: +1-760-731-3000
On Wed, 12 Jan 2011, Steve Edwards wrote:
On Wed, 12 Jan 2011, Gary Kuznitz wrote:
I currently have in extensions.conf:
exten = 106,1,Set(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)})
exten = 106,n,Monitor(wav,${CALLFILENAME},m)
exten = 106,hint,SIP/106
exten = 106,Macro(stdexten,106,${HINT
On 10 Jan 2011, at 10:17, Phuong Hoang wrote:
Thanks enkillar, but this is`nt thing that i need. I want to check number
online, offline or unreachable on asterisk using AMI(Asterisk Manager
Interface) by java but i have`nt found a solution yet. I hope you can help me
do this.
Thanks in
On 10 Jan 2011, at 10:37, Phuong Hoang wrote:
I found the link you have just sent to me but it do`nt help me to resolve this.
Can you say clearlier for me?
Not really. It's a list of manager commands. There is 'SIPshowpeer' which will
work for sip stuff. Try the command 'Command' action and
) The Asterisk source code. Even if you aren't a C programmer grepping
through the source code can be very productive.
--
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline
for testing when we
implemented G.722.1.
One of the annoying things about the Polycoms is trying to work out what
they can do. You have to search quite hard to find which codecs each
model supports.
Steve
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On 01/06/2011 05:25 AM, Tim Panton wrote:
On 5 Jan 2011, at 13:07, Steve Underwood wrote:
G.722.1 is a 7kHz bandwidth codec. G.722.1C is a stretched version offering
14kHz bandwidth. These are most often found in Polycom phones, but they are
available elsewhere. The only widely supported HD
bandwidth codec. G.722.1C is a stretched version
offering 14kHz bandwidth. These are most often found in Polycom phones,
but they are available elsewhere. The only widely supported HD codec is
G.722. Pretty much anything offering wideband voice supports G.722.
Steve
On 01/06/2011 01:04 AM, Tilghman Lesher wrote:
On Wednesday 05 January 2011 07:07:10 Steve Underwood wrote:
On 01/05/2011 03:29 PM, Bruce B wrote:
Hi Everyone,
1- Are the Siren7 and Siren14 the G.722 HD voice codecs?
2- Are these codecs only for Polycom units or are they universal
across all
On 01/06/2011 12:05 AM, Kevin P. Fleming wrote:
On 01/05/2011 07:07 AM, Steve Underwood wrote:
On 01/05/2011 03:29 PM, Bruce B wrote:
Hi Everyone,
1- Are the Siren7 and Siren14 the G.722 HD voice codecs?
2- Are these codecs only for Polycom units or are they universal
across all other SIP
project or vaporware. It is backed by Nokia and Intel.
http://ofono.org/
It looks like with a little glue, chan_ofono could quickly bring real
cell phone capabilities to Asterisk or other voip platforms.
Just an FYI,
Steve Totaro
-- Forwarded message --
From: Pekka Pessi ppe
not specifically looked at the asterisk
wiki, but Google searches brought up lots of messages confusing the
fax operation of the echo canceler with the faxdetect= setting for
DAHDI/Zaptel.
Steve
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On 01/04/2011 09:53 PM, Kevin P. Fleming wrote:
On 01/03/2011 07:08 PM, Steve Underwood wrote:
On 01/04/2011 04:22 AM, Kevin P. Fleming wrote:
No. CNG tone is never used to affect the state of an echo canceller.
All G.168 compliant echo cancellers will respond to the CED tone
(generated
on the host CPU. 'faxdetect' is not
set in DAHDI, it's set in chan_dahdi.conf.
Steve
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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does.
Matching is facilitated by patterns.
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Newline Fax: +1-760-731-3000
,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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On 24 December 2010 15:44, Steve Davies davies...@gmail.com wrote:
On 23 December 2010 18:01, Steve Davies davies...@gmail.com wrote:
Hi Again,
I thought I had this sorted, but it appears that in a clean
environment I did not in fact fix it. There appears to be a bit of a
contradiction.
1
normally wins over
better in the real world.
G.729 annex B is an add on, providing VAD features. It may be used with
G.729 or G.729A. A separate entry in the SDP says whether this option is
supported.
Steve
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was on.
Any chance sipsak or sipp can handle this task? I reboot my aging SPA3K
using an http request via wget.
--
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Newline
On Sat, Dec 25, 2010 at 7:41 PM, dave george dgeo...@teletoneinc.comwrote:
Yes we have that set in logger.conf.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Ustinov
Sent: Saturday, December 25,
another thread about problem with iax and Asterisk 1.6.2
(rsa auth not working anymore). Are there some known problems with iax and
1.6 version of Asterisk?
Thanks for any hint
Not 100% sure, but I think there was a fix for IAX audio in
1.6.2.16-rc1 - Perhaps try that?
Regards,
Steve
On 23 December 2010 18:01, Steve Davies davies...@gmail.com wrote:
Hi Again,
I thought I had this sorted, but it appears that in a clean
environment I did not in fact fix it. There appears to be a bit of a
contradiction.
1) In 1.6.2.x, musiconhold requires DAHDI (which we have)
2) In 1.6.2
On 7 December 2010 17:47, Steve Davies davies...@gmail.com wrote:
On 7 December 2010 15:00, Steve Davies davies...@gmail.com wrote:
On 7 December 2010 14:17, Lee Archer lee.arc...@thebigword.com wrote:
Hi, try unloading res_timing_dahdi.so then trying again.
Lee
-Original Message
a report on
https://issues.asterisk.org and I'll get a patch uploaded pronto.
Please let us know the issue number once raised - I'd like to follow this one.
Regards,
Steve
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the default timeout is
20 seconds.
Cheers,
Steve
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On 21 Dec 2010, at 14:20, A J Stiles wrote:
Well, every Free and Open Source telephony system is using Asterisk (and
Linux) under the bonnet. The differences are in the user configuration
tools.
Uh, no?
S
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,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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for
'bind' in sip.conf.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
On 13 Dec 2010, at 14:25, Danny Nicholas wrote:
(god forbid) postal mail
Haha, I'm kind of tempted to write an app_cups module to print envelopes ;)
S
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?
How about:
sip show registry
sip show peer server
--
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Steve Murphy
ParseTree Corp.
57 Lane 17
.
Any more suggestions?
Many thanks,
Steve
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On 10 December 2010 16:45, Steve Davies davies...@gmail.com wrote:
Hi,
On one of our asterisk systems that is quite busy, we are seeing the
following from 'netstat -s':
Udp:
17725210 packets received
36547 packets to unknown port received.
44017 packet receive errors
17101174
On 10 December 2010 17:21, Shaun Ruffell sruff...@digium.com wrote:
On 12/10/2010 11:02 AM, Steve Davies wrote:
On 10 December 2010 16:45, Steve Davies davies...@gmail.com wrote:
Hi,
On one of our asterisk systems that is quite busy, we are seeing the
following from 'netstat -s':
Udp
On 10 December 2010 17:33, Steve Davies davies...@gmail.com wrote:
On 10 December 2010 17:21, Shaun Ruffell sruff...@digium.com wrote:
On 12/10/2010 11:02 AM, Steve Davies wrote:
On 10 December 2010 16:45, Steve Davies davies...@gmail.com wrote:
Hi,
On one of our asterisk systems
On Thu, Dec 9, 2010 at 10:31 AM, Daniel Tryba dan...@tryba.nl wrote:
You could use SIPVicious to run attacks on your own servers:
http://code.google.com/p/sipvicious/
Yeah, why not? All the criminals on the internet are using it, too! ;^)
I'm seeing 1-4 scans per day on the average. And
and interaction between sip.conf and
extensions.conf you will be in a better place to evaluate the merits of
using a GUI to create your dialplan or continue growing your own.
--
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Steve Edwards sedwa
On 7 Dec 2010, at 11:35, Jonas Kellens wrote:
When on a public server, I find this insecure.
Then secure it? Tie down by IP address, or some phones support the
username:password@ in a URL.
S
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the following. Any thoughts on whet to check next?
Thanks,
Steve
### Call comes in here and is answered
-- SIP/snom360-0d6f answered DAHDI/2-1
-- Executing [...@macro-set-moh-call:1]
GotoIf(SIP/snom360-0d6f, 0?done) in new stack
-- Executing [...@macro-set-moh-call:2] Set(SIP/snom360
On 7 December 2010 14:17, Lee Archer lee.arc...@thebigword.com wrote:
Hi, try unloading res_timing_dahdi.so then trying again.
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Davies
Sent: 07
.'
--
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Newline Fax: +1-760-731-3000
On 7 December 2010 15:00, Steve Davies davies...@gmail.com wrote:
On 7 December 2010 14:17, Lee Archer lee.arc...@thebigword.com wrote:
Hi, try unloading res_timing_dahdi.so then trying again.
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk
,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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videos and diagrams
that are specific to Asterisk, Vyatta, and all of your questions.
http://www.google.com/search?q=vyatta+asterisk+qos
Thanks,
Steve T
On Sun, Dec 5, 2010 at 1:36 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Dear Steve;
I am fully thanks for your advise and kindly help.
I am
no more answers for you since you are unwilling to try to answer them
yourself first.
Thanks,
Steve T
On Mon, Dec 6, 2010 at 4:10 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Dear Steve;
Really until now, I am not able to know if Vyatta has a DSL router
(hardware) that can be used to do the QoS
the firewall, everything is
blocked, so if you are going to use it as a firewall, get as many rules in
place as you can think of.
Thanks,
Steve T
On Thu, Dec 2, 2010 at 3:14 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Dear;
I understood that Vyatta is the solution for the QoS, but I am
,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
Un-top-posting...
On Sun, 5 Dec 2010, Thomas Perron wrote:
Any reason why I don't get audio on the channel after it rings and the
end user picks up.
exten = s,n,Dial(SIP/callwithus/44,120,A,(demo-thanks))
On Sun, Dec 5, 2010 at 8:34 PM, Steve Edwards asterisk@sedwards.com
On 3 Dec 2010, at 13:47, Rodrigo Lang wrote:
unansweredy = yes
Remove the extra y.
S
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On Wed, Dec 1, 2010 at 12:15 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 12/01/2010 01:05 PM, Steve Murphy wrote:
Hello,
I wonder if anyone else has noticed this.
I see a pair of calls to pipe() within the codec_g729a, and suddenly, I
have a leaked file descriptor that remains
over each other and
the quality was five by five, except for solar flares, sandstorms,
rain. Things beyond my control.
Thanks,
Steve T
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New to Asterisk
, the audio was five by.
Several people that work for Digium that will remain anonymous, have
said to only use IAX when absolutely needed.
You will also see people agreeing with me and others that have no issues.
I just use SIP.
Thanks,
Steve T
On Thu, Dec 2, 2010 at 9:27 AM, Mark Deneen mden
:0.0.0.0
unless you know what you're doing.
--
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
decoding MP3s over and over? If you decode the files to
[wav|ulaw|slin|xxx] the files will 'just work' and you'll have more cycles
for more fun stuff like handling calls.
--
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-
Steve Edwards sedwa
On Thu, 2 Dec 2010, Steve Edwards wrote:
What does 'sip show settings' show? The first 2 settings (1.6.2.5) should
be:
UDP SIP Port: 5060
UDP Bindaddress:0.0.0.0
On Thu, 2 Dec 2010, Gary Kuznitz wrote:
In sip.conf bindport = 5060
'Sip show settings' doesn't work
to be 'unixfied?'
What does 'hexdump -C sip.conf' look like?
Does commenting (';') out line 1 change anything?
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in
separate directories, but all on the same development box.
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On 1 Dec 2010, at 10:18, Michael Nausch wrote:
If I start asterisk 1.8 with service asterisk start or
/etc/init.d/asterisk start, I can't load chan_misdn.so
If I run asterisk 1.8 as root via asterisk -vvvc I can access my ISDN-card
and I be able to dial out to my PSTN provider! ;)
File
at my spec, you can:
svn co http://svn.digium.com/svn/asterisk/team/murf/RFCs
and look at the pdf there in that directory.
murf
Steve Murphy
ParseTree Corp.
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don't have to go
thru
any bother.
Just keep in mind that clever people can/will take advantage of the fact
that everything after an incoming call is transferred is lost to billing (as
an example).
murf
Steve Murphy
ParseTree Corp
Hello,
I wonder if anyone else has noticed this.
I see a pair of calls to pipe() within the codec_g729a, and suddenly, I have
a leaked file descriptor that remains until asterisk dies.
Now, maybe no-one sees this, mainly because I have no g729 licenses on the
machines where this happens. And
On 30 Nov 2010, at 09:28, bilal ghayyad wrote:
If I ran IAX in TCP port, and in case my network was having a lot of users
doing browse on the internet and downloading, so in that case and if the IAX
used TCP port, so the voice will be better than using UDP (because in TCP the
lost packets
On 30 Nov 2010, at 09:47, Michael Nausch wrote:
I tried to configure Asterisk 1.8 on one of my test-hosts.
I've installed from centos-asterisk.repo
(http://packages.asterisk.org/centos/$releasever/tested/$basearch/):
snip
[Nov 30 10:35:53] WARNING[7281]: channel.c:5353 ast_request: No
with
YouTube or whatever.
Thanks,
Steve T
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On Tue, Nov 30, 2010 at 1:00 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Tue, Nov 30, 2010 at 4:28 AM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
Can I run the IAX on TCP port instead of UDP port?
If I ran IAX in TCP port, and in case my network was having a lot of users
Un-top-posting...
On Sat, 27 Nov 2010, Giuseppe D'alessio wrote:
Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels.
On Sat, 27 Nov 2010, Steve Edwards wrote:
2) Write a script to do asterisk -r -x 'core show channels', parse the
output and do asterisk -r -x
of a channel. Are you suggesting the OP
write an AGI so he can call into his system to ask it to hang up all
channels?
--
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Newline
On Sun, Nov 28, 2010 at 12:24 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Sun, 28 Nov 2010, Silver Thorne wrote:
I have noticed lately that there have been several attempts to hack our
Asterisk server.
So, I am wondering if anyone has a firewall/IP tables statement that
keep out
,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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On 28 Nov 2010, at 22:26, dotnetdub wrote:
It could be an extension name Where is the error trapping if this is the
case.. Who writes this shit?
A dedicated bunch of volunteers who don't appreciate you being a dick about
bugs, which you report without so much as a log entry or a core
${CHANNEL_NAME} for
each channel.
3) Write a script to do #2 using AMI.
--
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760
,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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From: Steve Edwards asterisk@sedwards.com
I'm still not sure what you are asking for. Are you wanting to pick up a
SIP [hard|soft] phone, dial an extension and hear yourself talk? If so,
the relevant configuration files will be:
On Sat, 27 Nov 2010, Ali Khalfan wrote:
yes
,
-
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Newline Fax: +1-760-731-3000
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visible most of the time.
Regards,
Steve
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-committal in its definition, but I always assumed
that an empty string was still a valid category-name, so GROUP()=123
is as valid as GROUP(X)=123.
Could this be clarified?
Many thanks,
Steve
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On 24 November 2010 10:12, Steve Davies davies...@gmail.com wrote:
I am confused. In Asterisk 1.2 and 1.4, in the code there is an error:
Setting a group requires an argument (group name)
But the syntax is shown as: Syntax: GROUP([category])
The [category] square brackets indicate to me
it. Standard
sized FAX images are 1780 pixels wide.
Steve
On 11/20/2010 06:02 PM, Michael wrote:
Hi,
We played around with the different parameters of the tif files and
found that the issue was with the resolution.
Most files generated on the PC have a 200x200 resolution, but it seems
that FFA
On 18 Nov 2010, at 10:36, Phuong Hoang wrote:
I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but
not successful, Can anyone help me to do it?
How is this different to the other two posts? Please stop repeatedly sending
messages! If nobody replies you're probably not
On 18 Nov 2010, at 10:33, Phuong Hoang wrote:
I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but
not successful, Can anyone help me to do it?
Given that you haven't given any error messages, any logs, or your sip.conf, or
the manner in which it is not working
'
to your [some_context] context, perhaps the following will work. I
have not tried it:
[some_context]
include = parkinglotA
include = outboundcalls
[parkinglotA]
exten = t,1,Verbose(1|parking timeout!!!)
Regards,
Steve
as the console log from a call coming in on each channel
will help in assisting you in resolving this issue.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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'phone home' either without
permission or with the permission granted somewhere in the silly EULA that
nobody reads.
Am I missing something really obvious?
--
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Steve Edwards sedwa...@sedwards.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards Sent: Wednesday, November 17, 2010 2:45 PM
Would you object to a 'curl' request in the script that starts Asterisk
that sends your MAC address and Asterisk version number to Asterisk.org?
Not tracking your IP
' runs a daily script that emails a tarball of the current
configuration to me.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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,
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Newline Fax: +1-760-731-3000
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, and thus it can only happen once per day.
On the main server, if you type, dahdi show channels or whatever, do you see
the fourth span?
Your configuration should probably be signalling = pri_net and the new
server should be signalling = pri_cpe
Thanks,
Steve T
an In
Service.
-Jonathan
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,
Steve Totaro
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? Or is it just like a serial number for the scan? What?
Here's some examples:
2648061411
3190339404
2685608247
3358171034
2092652562
2206598858
Just trying to follow the advice: Know thy Enemy
murf
Steve Murphy
ParseTree Corp.
57 Lane 17
Cody, WY 82414
✉ m...@parsetree.com
☎ 307-899-5535
On 7 Nov 2010, at 20:59, Thomas Perron wrote:
I have installed Asterisk before w/ no issues but while trying today
(1.6.2.13 and centors 5.4) I receive the following at the CLI:
The configure script must be executed before running 'make'.
Please run ./configure.
Any
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