On 02/03/2010 02:03 AM, Olle E. Johansson wrote:
2 feb 2010 kl. 11.20 skrev BERGANZ Francois:
Hello all,
Does asterisk accept uri tel: instead of sip: ?
No, but I think it would be a good addition.
Why? Just curious.
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Alex Balashov - Principal
Evariste Systems LLC
Tel: +1 678
I don't know about 4xx, but 503 would be more benign for general/
miscellaneous errors than 603.
--
Sent from mobile device
On Jan 29, 2010, at 3:54 AM, Olle E. Johansson o...@edvina.net wrote:
Agree that the 603 is wrong. It hasn't caused me issues but I see
where it could. And it goes
, or a new call leg in the case of the
UA) because of this precise implication of 6xx-class final replies.
-- Alex
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On 01/28/2010 04:47 PM, Kristian Kielhofner wrote:
On Thu, Jan 28, 2010 at 4:23 PM, Alex Balashov
abalas...@evaristesys.com wrote:
It's also problematic because a 3261-compliant SIP proxy or UAC is not
going to attempt to reach the destination by alternate means (serial
forking in the case
On Sat, Jan 23, 2010 at 08:08:28AM +0100, Philipp von Klitzing wrote:
Hi!
I was wondering if you can use the base station as a outbound pots
connection for asterisk.
I currently have a tdm410 to do fxs/fxo ports and would like to get rid of
it, I used to use a spa3102, but it only
and the server is as below.
N900 SIP client-- OpenVPN -- Asterisk
The version of Asterisk in question is 1.6.0.18.
Any suggestions?
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too. Perhaps some
others.
As far as how to implement it, that is manufacturer-specific. Look on
voip-info.org for Polycom and paging if you want the Polycom-centric
answer. For other phones, it will be different.
--
Alex Balashov - Principal
Evariste Systems LLC
Tel: +1 678-954-0670
On Fri, Jan 22, 2010 at 05:06:17PM -0300, Andrew Latham wrote:
I have worked on many snom phones over the years I have never had
a snom phone go bad...
I have had about 10 in the last 12-18 months, I had 1 with a fault hand
set plug - the reseller replaced it. Other wise they have been
to pass through ISUP attributes. For that, you would need SIP-T.
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to control
the pots fallover from asterisk.
if not the siemans are there any other bases that would fit the bill ?
Alex
On Fri, Jan 22, 2010 at 07:14:56PM -0500, John Hurley wrote:
From my experience, unless you have another base station for sets you would
want to configure separately
Asterisk underneath; FreePBX and Trixbox are simply
administrative GUI layers that provide you with a different way to
manage the configuration files. Using Asterisk straight would require
that you edit them by hand.
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Tel: +1 678-954-0670
the AST_CONFIG and AST_CDR tables, there's nothing accessing
the voicemessages table.
Any idea where I can look next?
TIA
Alex
Checked by Hu-fw-yhman
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heard VOIP hackers call this inbargeability; it's the ability
to barge in to a playing audio clip.
I'm planning to use Lumenvox for the DTMF and voice recognition, BTW.
Not sure if that matters.
Many thanks to anyone who can lend me a clue about this,
--
Alex Balashov - Principal
Evariste
When passing arguments to applications you must use parentheses.
Try:
exten = _X.,3,DeadAGI(a2billing.php)
You can omit parentheses when calling applications with no arguments,
e.g.
exten = s,1,Answer
... but not when there are parameters.
--
Sent from mobile device
On Jan 1, 2010,
Fact.
On 01/01/2010 01:06 PM, Warren Selby wrote:
Also, shouldn't the .php script be located in /var/lib/asterisk/agi-bin?
On Fri, Jan 1, 2010 at 7:31 AM, Alex Balashov abalas...@evaristesys.com
mailto:abalas...@evaristesys.com wrote:
When passing arguments to applications you must use
On Tue, Dec 29, 2009 at 11:30:21PM -0500, C F wrote:
Before I start I am a Panasonic certified dealer AND I have installed
over 100 Asterisk systems that are in production.
That said for your application use Panasonic, DONT use Asterisk.
Use the Panasonic KX-TDA50G. Supports up to around 50
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boxes as
well), when the adsl drops it causes a ip loop (6to4 routing), which
hoggs irq's this in turn causes a software bug in the dahdi/zaptel
driver which means I have to reload the dahdi/zaptel module in asterisk
- easy to capture (do it with ppp-up)
Alex
On Tue, Dec 22, 2009 at 11:53:02AM
is, the scalability requirements, the interface
requirements, etc.
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From: Alex Bell voicese...@gmail.com
Date: Sat, Dec 19, 2009 at 5:57 AM
Subject: Nortel BCM - Call Accounting Interface?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Dear List,
Need to know if anyone
for this...?
Thanks for your response,
Al
On Mon, Dec 21, 2009 at 8:27 AM, Steve Howes steve-li...@geekinter.netwrote:
On 21 Dec 2009, at 12:04, Alex Bell wrote:
Dear List,
Need to know if anyone on this list has had any experience
with using the Nortel BCM 50 for Call Account
Neither. Use call files or the AMI Originate command.
--
Sent from mobile device
On Dec 21, 2009, at 8:04 AM, Thomas Perron thomas.per...@gmail.com
wrote:
I want to have Asterisk Dial individual numbers and play a recording
if each answers.
If they don't answer then the code rolls to
are glad to help so long as you ask
an intelligent question, such as yours, just word it properly.
http://www.tek-tips.com/threadminder.cfm?pid=1361
Thanks,
Steve T
On Mon, Dec 21, 2009 at 8:53 AM, Alex Bell voicese...@gmail.com wrote:
Steve,
The * is the first step in moving a small 3
to pay
though the nose for it.
Al
On Mon, Dec 21, 2009 at 9:52 AM, Adam Tauno Williams
awill...@opengroupware.us wrote:
On Mon, 2009-12-21 at 13:27 +, Steve Howes wrote:
On 21 Dec 2009, at 12:04, Alex Bell wrote:
Dear List,
Need to know if anyone on this list has had any
be wrapped inside a custom OCF
resource agent script, if you're using Heartbeat v2.
--
Alex Balashov - Principal
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Dear List,
Need to know if anyone on this list has had any experience with using
the Nortel BCM 50 for Call Account Reporting using an IP connection to a
Linux / Asterisk interface? Presently, I have a BCM 50 installed that uses a
local Lenova Small Form Factor PC with a windows XP / os
El 18/12/09 11:31, Bruce Nik escribió:
I am amazed that there is absolutely no proper documentation on how to
connect to Asterisk AMI with PHP. All tutuorial just mention: pass Action:
originate Channel: SIP/1234, blah blah blah and never give a simple example
of php.
On Tue, Dec 15, 2009 at 08:33:56AM +0100, hbk wrote:
IAXDIAL is free on app store works great on WiFi even true NATs but seem
blocked for GPRS.
ta
HB
[snip]
Well I have a 3gs - will tell you how that goes.
installed (non cracked), but I am on wifi now, easy to configure and
On Tue, Dec 15, 2009 at 09:14:16PM +1100, Alex Samad wrote:
On Tue, Dec 15, 2009 at 08:33:56AM +0100, hbk wrote:
[snip]
My only concern with it - it's not just a voip client, its many other
things as well. not sure if I want to be a fring user as well as all the
other memberships I have
On Tue, Dec 15, 2009 at 08:59:34PM +0100, Benny Amorsen wrote:
Gavin Spurgeon gspurg...@dageek.co.uk writes:
iSip (£2.39)
http://itunes.apple.com/gb/app/isip-push-service-formerly-sipphone/id298202722?mt=8
I have been very impressed by the audio quality from iSip, at least from
the
On Mon, Dec 14, 2009 at 07:37:08AM +, Brian Chamberlain wrote:
Fring, it's free and works perfectly with an Asterisk server..
thanks
On 13 Dec 2009, at 10:15, Alex Samad wrote:
Hi
Got a new iphone, want to know about peoples experience with any apps
that work well
-- not down here in
the southeastern US, as far as I can tell. So I can't speak to
whether voice works over 3G.
--
Sent from mobile device
On Dec 14, 2009, at 6:57 PM, Alex Samad a...@samad.com.au wrote:
On Mon, Dec 14, 2009 at 07:37:08AM +, Brian Chamberlain wrote:
Fring, it's free
played with it since the last firmware
update though as the update removed support for 3rd party headsets .
On 12/14/09, Alex Balashov abalas...@evaristesys.com wrote:
I personally have not had much luck with these softphones because the
iPhone 3G seems to be underpowered and just doesn't
Hi
Got a new iphone, want to know about peoples experience with any apps
that work well with asterisk and run on a iphone
Alex
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On Dec 12, 2009, at 9:50 PM, Landy Landy landysacco...@yahoo.com
wrote:
Hi List.
Don't know if I already posted about this problem but, if I have I
apologize for the double post.
I am trying to test a time of day extension dialing
to whitespace.
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On 12/12/2009 11:54 PM, John Novack wrote:
Alex Balashov wrote:
On 12/12/2009 10:59 PM, Steve Edwards wrote:
A perfect example of Asterisk's asinine handling of whitespace.
Either that, or a perfect example of user imprecision. It's OK to demand a
certain grammar from the users of what
to think about it in a voice lab
for my studies.
Kind regards,
Vitor
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On 12/10/2009 02:07 AM, Tzafrir Cohen wrote:
Why would one want a daily sabotage of the system in the first place?
An apt question.
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Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
It would depend on the phone's support for various custom headers for
this purpose; there is nothing universal.
Otherwise, calling name is your best bet.
--
Sent from mobile device
On Dec 7, 2009, at 3:00 AM, Giedrius Augys voi...@gmail.com wrote:
hello,
I've callcenter and our queue
Thorolf,
In the [general] section of sip.conf, set the 'bindaddr' parameter to
the cluster IP. If Asterisk is only bound to the floating interface, it
will respond only from that source IP.
-- Alex
Thorolf Godawa wrote:
Hi all,
I installed a Linux-HA-cluster with DRBD and Asterisk 1.4
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-INVITEs, BYEs,
etc.) can be routed based on the Route header even if the runtime
transaction state has been lost.
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Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
and simulated process space isolation. In
the latter, an actual virtual machine is run, although it is still
native-bound to some degree in that it is not a pure
userspace-scheduled process.
So, the answer is somewhat qualified; it depends on what is meant by
generalisation.
-- Alex
--
Alex
happens if you want
to play recorded messages and things. It would probably need licenses then
because its encoding.
Sent from my Windows Mobile® phone.
-Original Message-
From: Alex Balashov abalas...@evaristesys.com
Sent: 02 December 2009 01:13
To: Asterisk Users Mailing List
Steve Howes wrote:
On 1 Dec 2009, at 09:20, Benny Amorsen wrote:
I do believe that we have run out of brackets...
Could always do it vertically and useV and ^ ;)
And before you know it, we'll be here:
http://www.catonmat.net/blog/secret-perl-operators/
:)
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Alex Balashov
-users
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?
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question to begin with, or considering their problem in a
larger context.
Jeff LaCoursiere wrote:
Next question will be How can I keep my server from crashing? :)
(add more RAM... which may have been a good answer for question 1...)
j
On Tue, 24 Nov 2009, Alex Balashov wrote:
Disable swap
://lists.digium.com/mailman/listinfo/asterisk-users
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I am talking about the endpoints (extensions). SIP? DAHDI? IAX? H.323?
das sandesh wrote:
Hi Alex,
I am using Ring All channel strategy...
Thanks
Sandesh
On Tue, Nov 24, 2009 at 10:21 AM, Alex Balashov
abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote:
What
. The option to remove it is
contingent upon refraining from use of dial plan applications that
implicitly invoke it.
-- Alex
Ishfaq Malik wrote:
Hi
All my incoming dial plans start of with an Answer which I now know
starts the billing time. Some of the dialplans then get forwarded out
wrong with the PSTN.
DNID breakage is a long-standing Asterisk problem.
If this is taking place in the context of the dial plan, why not just
use ${EXTEN}?
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Alex Balashov - Principal
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Tel : (+1) (678) 954-0670
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I am curious what happens if you do the following instead:
[abc]
exten = _.,1,Answer
exten = _.,n,NoOp(${EXTEN})
ABBAS SHAKEEL wrote:
Thanks Alex,
suppose this is the context
[abc]
exten = s,1,Answer();
exten = s,n,Noop(${EXTEN});
exten = s,n,Noop(${CALLERID(dnid)});
I get
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,Hangup
[context_2]
exten = 6789540671,1,Dial(SIP/abalashov,30,r)
exten = 6789540671,n,Congestion
-- Alex
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Danny Nicholas escribió:
Could it be your using option X when you have no extensions for the user to
exit to (therefore when they press dtmf instead of one and done, they just
keep going?)
_
From: asterisk-users-boun...@lists.digium.com
If you are referring to VoiceXML, there is no (open-source) VoiceXML
environment for Asterisk at this time.
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Kevin P. Fleming wrote:
Alex Balashov wrote:
If you are referring to VoiceXML, there is no (open-source) VoiceXML
environment for Asterisk at this time.
Indeed there is: http://www.voiceglue.org/about/
I stand surprised and corrected.
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Web
On Tue, Nov 17, 2009 at 09:09:39AM -0800, Steve Edwards wrote:
On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote:
Hi All,
[snip]
2. Run from the external shell prompt:
asterisk -rx 'help whatever' | less
Or, you can use the script command to capture the output to a file
On Mon, Nov 16, 2009 at 08:17:27AM -0600, Kevin P. Fleming wrote:
Alex Balashov wrote:
As far as I know, Asterisk has no way to restrict the content of the
domain portion of the Contact URI. However, most commercial SBCs
should have a way to filter this, and it is highly recommended
piping an internet stream into a phone call via some
app
in an extension?
Alex Balashov abalas...@evaristesys.com wrote:
cov...@ccs.covici.com wrote:
Is there any app to pipe a stream to a call either a meetme
conference
or even a regular call?
Do you mean piping outside audio
I suppose that would depend on how the information about the
registrations is organised; do you want Asterisk to query some sort
of database used for backing these registrars and figure out where the
contact binding for a given AOR resides? AGI and func_odbc provide
fine ways to do that.
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it this way: if everyone did what you just did, the
mailing list would be almost useless.
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trivially with the use of a well-placed AMI
Originate command (or, perhaps, call files) combined with Local dial
plan channels.
-- Alex
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contexts, where the
input files come from, etc?
MoH doesn't get generated magically.
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-Install SOX (Sound Exchange Quality)
yum install sox
2 install a music (plz give the link to download)
3 convert the song in asterisk format, convert it through sox
4 file saved in mohwav
5 now give the path in /etc/asterisk/musiconhold
thx
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Thanks. In that case, do me a favour in return and start using the
mailing list as it is intended, instead of mailing people privately.
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On Nov 14, 2009, at 1:58 PM, aster...@opensourcesolution.in wrote:
hi alex done with music on hold. n thanks a lot
.
Cheers
On Sat, Nov 14, 2009 at 11:56 PM, Alex Balashov abalas...@evaristesys.com
wrote:
I don't get it. I just replied helpfully to Mr. opensourcesolutions
on the mailing list and for this he expresses his gratitude with two
more obnoxious private e-mails to me, along these general lines
, someone else new to
Asterisk who is in your position later on - but perhaps slightly more
resourceful than you are - can search the list archives on Google and
benefit from this discussion.
As far as your question, it seems to me that you did everything right.
-- Alex
aster...@opensourcesolution.in
laziness and respect, not legitimate differences
of language, culture and technical abilities of which we should all -
I fully agree - be mindful.
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, expressive laziness and lack of
consideration for the charity of one's audience at best, and brain
damage at worst.
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What say you to the proposal that some approaches to seeking help are
so ridiculous they should not be tolerated?
Community standards neither conceive nor enforce themselves.
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Sent from mobile device
On Nov 13, 2009, at 2:32 PM, Cary Fitch ca...@usawide.net wrote:
Slightly paraphrasing a
.
I've seen people with far, far worse English than our
opensourcesolutions friend interact in a way that reflects much more
common sense, awareness, respect of the audience, and a generally
cultured way of seeking assistance that dignifies a response.
-- Alex
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he posts a brilliant tome Subjected installing
consisting of 2 words -- installing asterisk. He received less than
helpful responses from Steve Howes, Alex Balashov, and Pascal Bruno.
Date: Wed, 28 Oct 2009 14:07:30 +
Subject: [asterisk-users] deploying asterisk
Pawan states he had just
: Reduce the frequency with which the word solution appears in
your sentences by about ... 5000%.
-- Alex
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product, and rather similarly at that.
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Direct : (+1) (678) 954-0671
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Try aretta.com.
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On Nov 11, 2009, at 6:41 AM, Paulo Vicentini vizent...@hotmail.com
wrote:
Hi,
I am looking for a hosted / virtual IPBX *PLATFORM* for service
provider.
Such hosted IPBX platform is aimed to be as a service, so that final
customers don't have
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asterisk-users mailing list
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Alex Balashov
/asterisk-users
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Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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asterisk-users mailing
be made by the sending user agent to that
destination. This is not true of 4xx and 5xx-class errors.
Also, why is your name rendered in all-capital letters? Have you
considered becoming Dhaval Indrodiya instead of DHaVAL INDRODIYA?
-- Alex
--
Alex Balashov - Principal
Evariste Systems
Web
The problem is with your provider, unless there is something wrong
with about 1/4th of your calls - i.e. the destination is unroutable by
that provider.
DHAVAL INDRODIYA wrote:
thanks Alex,
thanks for your reply,
is there any changes needed for resolving this issue , in sip.conf
://lists.digium.com/mailman/listinfo/asterisk-users
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Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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it works.
IAX2 appears to have a text frame type:
static int iax2_sendtext(struct ast_channel *c, const char *text)
{
return send_command_locked(PTR_TO_CALLNO(c-tech_pvt),
AST_FRAME_TEXT,
0, 0, (unsigned char *)text, strlen(text) + 1, -1);
}
-- Alex
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Alex Balashov
a message.
-- Alex
Hakan C wrote:
It does nothing on hardware channels.
SendText is just works on SIP channels.
Purpose of SendText is showing text messages on user phone screen.
show application SendText
-= Info about application 'SendText' =-
[Synopsis]
Send a Text Message
.
That is correct.
It sounds like your need to make sure you're using the same trunk
group within DAHDI over and over:
Dial(DAHDI/1/${LOCAL_DIAL})
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
windows machine.
now i want to install softphone in both windows machine. and both
softphone should communicate with each other. any support and guidance
will be highly appreciated.
thx
--
Alex Balashov
he is in the wrong business.
Steve
On 9 Nov 2009, at 16:32, Alex Balashov wrote:
As I said, please keep discussion on list.
aster...@opensourcesolution.in wrote:
first of all i appologise for sending u pvt email. i have installed
asterisk on Centos 5.3, plz open the attachment in which
You just don't get it, do you?
Your indolent methods of getting what you want are not at your
disposal here.
This is not a homework help forum.
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Sent from mobile device
On Nov 9, 2009, at 12:11 PM, aster...@opensourcesolution.in wrote:
hi all,
i have installed asterisk on Centos 5.3,
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