already built your
intended product, and rather similarly at that.
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keting and communication advice, which you can take or
leave: Reduce the frequency with which the word "solution" appears in
your sentences by about ... 5000%.
-- Alex
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Try aretta.com.
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Sent from mobile device
On Nov 11, 2009, at 6:41 AM, Paulo Vicentini
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 to install, maintain,
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 ,
onses indicate that no further
call attempts should 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
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/listinfo/asterisk-users> for both
>> calls. I have a Variable />/ with the LOCAL_DIAL set.
>> /
>> That is correct.
>>
>> It sounds like your need to make sure you're using the same trunk
>> group within DAHDI over and over:
>>
>>Dial
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Web
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.
--
Sent from mobile device
On Nov 9, 2009, at 12:11 PM, wrote:
> hi all,
>
> i have installed asterisk on Centos 5.3, plz i had installed one
>
phone. Clearly 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 e
isk machine and two 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
>
>
> ---
y_context for both calls. I have a Variable
> with the LOCAL_DIAL set.
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})
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"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 'SendTe
t exactly the applicable standard is or how 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
0109)
>
>
>
>
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On Nov 8, 2009, at 2:13 PM, "B.Masoud @ SH" wrote:
> I have 2 questions:
>
>
>
> 1. Can I make outbound route rule based on the Source Channel?
>
Yes.
> 2. Can I auto change the outbound route based on time/Day of
> week?
>
Yes. See GotoIfTime().
>
asterisk -v
> bash: asterisk: command not found
> [r...@localhost etc]# asterisk -V
> bash: asterisk: command not found
>
> thx
>
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XO card. TDM hardware that
interfaces with T1/E1 circuits (ISDN PRI, typically) is also available.
-- Alex
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sive work on one's behalf to fix a problem one has not done
the due diligence to rudimentarily understand.
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In a manner of speaking.
DHAVAL INDRODIYA wrote:
> is this answer of my question?
>
> i cant understand?
> regards
> Dhaval
>
> On Mon, Nov 2, 2009 at 12:25 PM, Alex Balashov
> mailto:abalas...@evaristesys.com>> wrote:
>
> Dhaval,
>
> Why
e
>
> very long.
>
> regards
> Dhaval
>
>
>
>
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assisting Marriott customers).
> Perhaps
> wherever you are located the state of firewalls/routers is different.
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex
> Bala
often the only way our staff can connect while on the road.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
> Sent: Friday, October 30, 2009 8:03 AM
> To: Asterisk Us
Vincent wrote:
> Since SIP/RTP is a pain to use with road warriors who need to connect
> from any location over the Internet, I'd like to get them some IAX
> phones instead.
What gives you that idea?
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Web : http://www.evaris
gh this project.
>
> 3) Break out the phonebook and start looking for a good lawyer. Being on
> the wrong side of a lawsuit is not pleasant.
>
> 4) Pick up a copy local paper and check out the want-ads.
>
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BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces, timer
> Content-Length: 0
>
> Ok, i see that:
>
> 1- Cisco sent the phone number of the caller (47700)
> 2- I have a "To: "
>192.168.50.130 = My Asterisk Server
>192.168.50.125 = My Cisco AS5300
>
routing hop cannot be found
out because the nodes are in a different MAC domain.
NAT has absolutely nothing to do with this, and thus is irrelevant one
way or another.
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Direct : (+1
600,3,Dial(SIP/Jpc,25,m)
> exten => 042600,4,Hangup
>
>
>
> And second problems:
>
> "Call from '' to", AS5300 don't sent the number of the caller ?
>
>
> Thanks for your help
> Jerome
>
>
>
>
>
>
:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Liivo,
I wonder if you are dealing with this general class of issues:
https://issues.asterisk.org/view.php?id=11491
-- Alex
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Tel : (+1) (678) 954-0670
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aster...@opensourcesolution.in wrote:
> installing asterisk
I am intrigued by your ideas and would like to subscribe to your
quarterly newsletter, as well as attend your biannual leadership seminar.
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Web : http://www.evaristesys.com/
dvise?
>
> Regards
> Bilal
>
>
>
>
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Ping.
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On Oct 24, 2009, at 8:33 PM, Matt wrote:
> This is a test... I am being told I am subscribed, but I am not
> getting messages.
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>
> asteris
> one of the less reliable methods.
The scope of the underlying intent behind ACk was a bit broader than that.
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>http://lists.digium.com/m
rovide some overview of what the
"operations" are. Few people will happily sign up to support
something they do not have any conception of.
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date options visit:
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activated to be more informational
send failure: Connection refused
Or it would just time out.
This command returns a value which can be captured by a shell script,
and so it is possible to build it into something like a Nagios plugin,
for example.
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es
> out at 24T1s, just shy of what you can get out of the Adtran MX 2800.
Yes, the AS5300 chassis can only do 4 T1s. You're looking for an
AS5400, or another big router chassis that can take a DS3 adaptor and
VFCs (like a 7200).
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Web
-4332 x105
>
>
>
>
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Interesting on my asterisk box I have installed Virtualbox and I run my
firewall/router in a vm, stripped down linux box with iptables, I have
snapshoted the image to a working image. it only does ip
forwading/vpn/iptable stuff ends up being a low foot print, 256M + 8G /
Alex
On Tue, Oct 13
asterisk server and just treat it as a extension, but letting all
its outbound calls go through the pstn line on the tdm card - will the
fax work that way ? the question is still how to I deal with inbound
calls/faxes ?
Alex
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Alex
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To
stricon.net
>
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If not, it will solve many other problems you would otherwise have.
ABBAS SHAKEEL wrote:
> Thanks Alex
>
> By just avoiding this will solve this problem?
>
> On Sat, Sep 26, 2009 at 9:47 PM, Alex Balashov
> mailto:abalas...@evaristesys.com>> wrote:
>
>
>
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On Thu, Sep 24, 2009 at 05:32:24PM -0500, Michael Graves wrote:
> On Thu, 24 Sep 2009 09:42:25 +0100, Steve Davies wrote:
>
> >Hi,
> >
> >Given that the Digium transcoding card has no external connections
> >(AFAIK), it strikes me that it would suit a mini-PCI slot very well.
> >
> >Does such a be
On Wed, Sep 23, 2009 at 09:39:09AM -0700, mgra...@mstvp.com wrote:
> I had a good experience with that Polycom/Spectralink phone. Very rugged
> as you say. The experience did highlight the weaknesses in consumer
> Wifi AP, which reinforced my commitment to continue using DECT around my
> office.
On Tue, Sep 22, 2009 at 07:57:56AM -0400, Leif Madsen wrote:
> Alex Samad wrote:
> > 4. GotoIf([${DIALSTATUS} = CHANUNAVAIL]?pt:ok)
> > [pbx_config]
>
> > i believe i have captured the relevant logging from the console. my problem
>
ll also
be necessary - not optional - if you plan to accept incoming calls
from the same carrier, so that you can set it not to challenge the
carrier for authentication credentials:
insecure=port,invite
and to route the calls into a particular dial plan context:
cont
@dial-sipmnf-sippt-pstn:4] GotoIf("DAHDI/1-1", "[BUSY =
CHANUNAVAIL]?pt:ok") in new stack
-- Goto (dial-sipmnf-sippt-pstn,s,5)
from my understanding BUSY != CHANUNAVAIL, therefor it should have jumped to ok
which is s,9.
What have I missed !
thanks
Alex
signatu
it's at for instant provisioning changes.
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On Thu, Sep 17, 2009 at 12:02:16PM +0300, Tzafrir Cohen wrote:
> On Thu, Sep 17, 2009 at 08:18:13AM +1000, Alex Samad wrote:
> > Hi
> >
> > how do i set the call-limit on a dahi line - its connected to the pstn
> > network - shared fax line. How do i tell asterisk
You can set some kind of counter in the dial plan, call an AGI script,
use func_odbc to make database calls, or otherwise achieve this
programatically.
--
Sent from mobile device
On Sep 17, 2009, at 1:16 AM, Patrick
wrote:
> Hello guys,
>
> I've one SIP trunk that support multiple DID. On
Hi
how do i set the call-limit on a dahi line - its connected to the pstn
network - shared fax line. How do i tell asterisk not to send more than
1 call there !
Alex
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St.
On Wed, Sep 16, 2009 at 12:24:22PM -0700, Steve Edwards wrote:
> On Wed, 16 Sep 2009, Danny Nicholas wrote:
>
> > I'd try this:
> > - exten => 4000,1,Dial(SIP/4000,20,ikKtT)
> > - exten => s-NOANSWER,1,Dial(SIP/4001,20,ikKtT)
> > - exten => s-NOANSWER,2,Voicemail(4000)
> > - exten => s-BUSY,1,Dial
hatever the
justification) will always get you. :-) I was pouring over the capture
you provided, thinking, "This really doesn't add up..."
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a.
Would you mind pasting a capture of the transaction in question, from
the vantage point of the outside interface of your Asterisk host? You
can change the representations of the external IP to something else if
you don't want to post it to a public list.
Thanks,
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-digital.com --
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table while im
> awake these days
just for clarity I am running on 1.6
[snip]
Alex
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R
G1},${ARG2})
exten => s,11,Set(GLOBAL(FOUNDME)=${DIALSTATUS})
exten => s,12,Goto(s-${DIALSTATUS},1)
;
Alex
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Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-
The Safisystems Team:This is an awesome app. I'm downloading
it now and will let you know my experiences.
Thanks for all your efforts to the OS Community,
Al
*
*
On Wed, Aug 26, 2009 at 2:48 AM, zac wolfe wrote:
> After over a year of alphas, betas, and release candidates I'm ha
I'm not sure I accurately understood the problem, but it sounds like
this is a failure to use non-blocking I/O on the client?
Raimund Sacherer wrote:
> oh boy,
>
> On Aug 25, 2009, at 10:05 PM, Alex Balashov wrote:
>
>> I've never seen that, myself. But I
Jason,Echo Hellooo old buddy! Do you evefr go 4 the dCap?
Your bud from the MD class.
Al
On Tue, Aug 25, 2009 at 4:43 PM, Jason Baker wrote:
> I recently upgraded my Asterisk system to Dahdi and now I have an echo
> problem.
>
> I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a D
On Tue, Aug 25, 2009 at 07:30:08PM +0200, Olle E. Johansson wrote:
>
> 25 aug 2009 kl. 18.50 skrev John A. Sullivan III:
>
> > On Tue, 2009-08-25 at 18:28 +0200, Olle E. Johansson wrote:
> >> 25 aug 2009 kl. 16.20 skrev Olivier:
[snip]
> > mode
> > in Linux on any old switch and it works reason
ity quirks - must be
properly managed. More boxes to spread the calls onto and
underutilising the hardware on each node is a better extreme to tend
toward than the opposite.
-- Alex
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Tel : (+1) (678)
results via AMI.
It's a question of ROI on your time. You can do it however you want,
especially if you're really motivated to avoid a particular type of
development chore.
-- Alex
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Tel : (+1) (67
Sanjoy Rath wrote:
> I would prefer to use AMI. Let me start looking into AMI. I would like
> to include functionalities like upload numbers to call from an
> interface, i want reports back numbers called, setup call time etc. Let
> me look up AMI. Thanks Alex for the info.
Yep.
sterisk-users-
boun...@lists.digium.com] On Behalf Of Sanjoy Rath
Sent: Tuesday, August 25, 2009 10:48 AM
To: Asterisk-Users
Subject: Re: [asterisk-users] Asterisk Autodialer
Alex,
You are right. My questions are probably wide open. I probably would
have more specific. Any help I get would be
ase, practically any modern distribution will do. Asterisk is
run on Debian, Ubuntu, CentOS, Fedora, RHEL, SuSE, and many more.
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extension in a
>>> SER that I can pass on as I've never looks at one?
>>>
>>> Cheers
>>>
>>> Ish
>>>
>>>
>>
>>
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On Fri, Aug 21, 2009 at 08:53:23AM -0400, Steve Totaro wrote:
> On Fri, Aug 21, 2009 at 8:39 AM, Alex Samad wrote:
>
> > Hi
> >
> > I had a working system, until recently - its asterisk 1.6.1 from debian
> > - not the lastest as the last doesn't seem to wor
just as bad, now I am not sure what to
check to try and get this working again .....
Alex
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t use MP3. Why would you want to burn CPU cycles decompressing the
> same stuff over and over?
Yep, agreed. Convert the file to the native codec(s) in which it will
be played.
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On Tue, Aug 18, 2009 at 10:00:55AM -0400, Dave Fullerton wrote:
> Kevin P. Fleming wrote:
> > Jeff LaCoursiere wrote:
> >> On Tue, 18 Aug 2009, Kevin P. Fleming wrote:
> >>
> >> [snip]
> >>
[snip]
>
> Here's my $0.02. If you don't want an echo canceller, specify
> echocanceller=none,x-y and hav
e ? I
> have tested Flite , cepstral (i have not buyed lisence for it trial
> only) but still thinking may be i have a good option ?
There are many good TTS engines; the best ones are, in my opinion, not
directly associated with Asterisk -- for example, Voxeo's stuff.
--
Alex Bala
that in Java without using Asterisk-java?
>
> Kind regards,
>
> Olivier
>
>
> Stefan Reuter a écrit :
>> Yes this is a well-known change. If you use the latest Snapshot of
>> Asterisk-Java 1.0.0 (http://asterisk-java.org) the issue should be resolved.
>>
short lifed ipv6 routing looping - sucks up cpu and doesn't let it intr
which starves the other cards.
Alex
> [] intel_machine_check+0x0/0x146 []
> error_code+0x39/0x40 [] mwait_idel+0x25/0x38 []
> acpi_processor_idle+0x154/0x3b4 [] cpu_idle+0x9f/0xb9
> ===
>
s, log entries, what the Asterisk CLI says
when set to high verbosity, etc.
--
Alex Balashov - Principal
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
___
--
Insufficient information.
Tareq Kibria wrote:
> Dear Alex ,
>
>Definitely there is reason behind it which i dont
> understand properly...I am novice to the system.
>
> For Inbound calling i create ingroups and assign it to the
> campaigns...USER logge
icon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
>
> Any ideas ?
>
> regards
>
> harry.
>
>
>
>
> ___
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>
>
Shashi Dookhee wrote:
> The question is, would it be more reliable to offload all dahdi/zaptel/libpri
> type stuff to a dedicated gateway device (Asterisk or Cisco)
Yes - for Cisco. No - for Asterisk.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1
Nowhere does the letter say Twitter is suing you. It is a cease and
desist letter.
I suppose their threat about further action at the bottom can be
reasonably surmised to mean that they might sue you in the future, but
that is a far, far cry from "Twitter is suing me!"
--
Ale
n be coaxed into putting out RADIUS
CDRs that CDRTool can operate on. It's RADIUS, after all.
But:
http://www.ag-projects.com/content/view/51/96/
"CDRTool is an Open Source solution that provides mediation, accounting
and tracing for Call Detail Records generated by OpenSER by using
w.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Alex Balashov
Evariste Sy
Word of advice: When you try SIP clients, focus on how the far-end is
hearing you, not whether you can hear them. In my experience, that's
where 90% of the deal-breakers lie with the iPhone.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954
Try prefix your extension in extensions.conf with "_", e.g.
exten => _123,1,...
--
Sent from mobile device
On Aug 10, 2009, at 6:55 AM, Patrick Plattes
wrote:
> Hello,
>
> i have a problem with the context parameter in the sip.conf. i'm using
> a german sip provider (sipgate.de) and every
http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
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> To UNSUBSCRIBE or update options visit:
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--
Alex Balashov
Evariste
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