steve;
thanks a lot
mike
On Oct 10, 2008, at 2:20 PM, Steve Totaro wrote:
On Fri, Oct 10, 2008 at 5:55 PM, Brent Davidson <[EMAIL PROTECTED]
> wrote:
Babcock, Michael Alex wrote:
> hey;
> i'm at best western and am curious is there a way i could find out
if
> our be
no i'm a guest at the bestwestern
On Oct 10, 2008, at 1:55 PM, Brent Davidson wrote:
> Babcock, Michael Alex wrote:
>> hey;
>> i'm at best western and am curious is there a way i could find out if
>> our best western, with out asking, is using asterisk?
>
don't know whether to look in the settings on the phone or in an
> Asterisk setting, and what setting to check in either place. Has anyone
> seen this behavior before?
> --Paul
>
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954
hey;
i'm at best western and am curious is there a way i could find out if
our best western, with out asking, is using asterisk?
oh and petsmart i think is using asterisk they have alason voice for
there main voicem enu.
mike
thanks for reading
Systems administrator and owner of http://gwhost
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>
at it. Maybe we can figure it out
> for you and add it to the wiki for everyone.
>
> One thing I don't want to do is duplicate effort elsewhere,
> copy/paste from other sites, etc. If you can link to an external
> resource, please do!
>
> In case you missed it befo
ues in the configs?
>
> thanks
>
>
> On Oct 9, 2008, at 9:36 AM, Alex Balashov wrote:
>
>>
>> This is due to an SDP mismatch of some sort, codec or otherwise.
>>
>> Perhaps you have not set your Asterisk SIP peers to support RFC2833
>> DTMF? Try dtm
okups on outbound
> calls
>
> [as5300_1]
> type=peer
> host=172.31.2.7
> permit=172.31.2.7/255.255.255.255
> defaultip=172.31.2.7
> disallow=all
> allow=ulaw
> allow=gsm
> allow=alaw
> nat=no
> canreinvite=yes
> dtmfmode=rfc2833
>
> I have also included link
digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile
, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel, x-sipura
Content-Type: application/sdp
can any body help me to over ride the callerid?
Thanks,
Max Alex
Voip Developer
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and other languages
designed to be web scripting languages and whose state is expected to be
determined in terms of serial HTTP requests.
I use Perl, personally:
http://search.cpan.org/~jaywhy/Asterisk-FastAGI-0.02/lib/Asterisk/FastAGI.pm
-- Alex
--
Alex Balashov
Evariste Systems
Web: h
this might be classifyed to some of you as "ot" but i wonder what this
will do for the asterisk community?
Begin forwarded message:
Date: October 5, 2008 5:17:36 PM GMT-08:00
Subject: Fonolo: Visually Navigate & Dial IVR Phone Menus in Web/
Mobile Browser
Source: Tech[dot]Blog
Author: Abdul
Colocation Provided by http://www.api-digital.com --
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;> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
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>
> A
What does it do? I mean, for someone like you, practically speaking?
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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riginal Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Gordon
> Henderson
> Sent: Friday, October 03, 2008 6:03 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] sip clients for smart phones?
>
>
egister Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678
ck speaker.
>
> Thanks,
> Matt G
>
> : http://www.voipphreak.ca
> : http://www.ratemydialplan.com
> : http://www.asterisk-jobs.com
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Gordon
> Henderson
> Sent: Frid
sipp can simulate RTP traffic.
Jai Rangi wrote:
> Al and Alex,
> Thank you for your input,
> Sorry TDM is not the option at this time :( .
> Everything has been great until last 2-3 days. Machine loads is not the
> issue, we have multiple asterisk server to share the load. Not mu
I
deal with
> the back speaker.
>
> Thanks,
> Matt G
>
> : http://www.voipphreak.ca
> : http://www.ratemydialplan.com
> : http://www.asterisk-jobs.com
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Gordon
> He
erisk-jobs.com
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Gordon
> Henderson
> Sent: Friday, October 03, 2008 6:03 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] sip cli
i should have been more clear, thanks a windows mobile smart phone sip
client. sorry about that
On Oct 3, 2008, at 1:54 AM, randulo wrote:
> On Fri, Oct 3, 2008 at 10:08 AM, Babcock, Michael Alex
> <[EMAIL PROTECTED]> wrote:
>> curious does anyone know of a smart phone client
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>http
windows smart phone v 6.0 example
htc shadow
is what i have. It has wifi abilitys.
mike
On Oct 3, 2008, at 12:11 AM, Alex Balashov wrote:
> What's a smart phone?
>
> Babcock, Michael Alex wrote:
>
>> hi;
>> curious does anyone know of a smart phone client that coul
much smaller than that MTU, or any
reasonable MTU you could set.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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What's a smart phone?
Babcock, Michael Alex wrote:
> hi;
> curious does anyone know of a smart phone client that could connect to
> asterisk?
> thanks
> Mike
> thanks for reading
> Systems administrator and owner of http://gwhosting.net
> msn: [EMAIL PROTECTED]
hi;
curious does anyone know of a smart phone client that could connect to
asterisk?
thanks
Mike
thanks for reading
Systems administrator and owner of http://gwhosting.net
msn: [EMAIL PROTECTED]
twitter: http://twitter.com/creepyblindy
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to functions and using that functions in dialplan.
but it is always gives me function is not registered.
can any body explain how to register custom functions in asterisk?
Thanks,
Max Alex
Voip Developer
___
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ortant those 120 users are.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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ng to cause media to cease in the
middle of a call, no.
--
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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Michael Collins wrote:
> IANAL but it looks like a lot of people have their hands out expecting
> payment for people using G.729: www.sipro.com, e.g.
Heh. :) Well, I have my hands out expecting a Treasury bailout...
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.co
Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>
>
>
>
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> AstriC
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Evariste Systems
Web: htt
t a good way to do real-time failover for anything.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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nstruction sets available to assist it, how
much memory is allocated to each VM, and other architectural
considerations.
Any perspective would be helpful, however.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mob
08 - September 22 - 25 Phoenix, Arizona
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Evariste Systems
Web: http://www.evaristesys.com/
Te
er is much harder unless
you are a carrier and have NPAC access.
Contact me privately off-list if you are having problems with a DID and
I might be able to help you determine the underlying carrier.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954
t; service in general and in the future. Their tech support has been
> absolutely cavalier to the point of insulting in refusing to deal with
> this basic issue of connectivity. I'm wondering if my experience is unique.
>> From: Alex Balashov <[EMAIL PROTECTED]>
>> It
BTW, if you provide the originating number, the underlying carrier can
be determined, either by the pooling or NANPA block it is assigned to,
or its LRN if ported. If you want, you can privately e-mail me the
number and I'll tell you who the carrier is.
Alex Balashov wrote:
> If Vit
itelity and other
> providers? Is there a way that I can determine whom to contact given
> only an originating number? Any words of wisdom? Documents I can read
> for educating myself?
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678)
right will stay away from them, smile.
mike
On Sep 29, 2008, at 9:01 AM, Gordon Henderson wrote:
> On Mon, 29 Sep 2008, Babcock, Michael Alex wrote:
>
>> what are 70 numbers?
>
> Prefix 070 (then 8 more digits) These are so-called "personal"
> numbers.
> They&
thanks for all this information.
michael
On Sep 28, 2008, at 11:37 PM, Gordon Henderson wrote:
> On Sun, 28 Sep 2008, Babcock, Michael Alex wrote:
>
>> hi;
>> i do not know how it works in the uk, but is there an equalivent to
>> our 866-877-888-800 numbers for london for
call for them - stay away from 070 numbers though.
2008/9/29 Babcock, Michael Alex <[EMAIL PROTECTED]>
hi;
i do not know how it works in the uk, but is there an equalivent to
our 866-877-888-800 numbers for london for say? I have some friends in
london and want them to be able to call me
om --
>
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hi;
i do not know how it works in the uk, but is there an equalivent to
our 866-877-888-800 numbers for london for say? I have some friends in
london and want them to be able to call me in the states.
Please help with where i can get the numbers, what they start with,
how much they are, and w
i'm using lylix, does anyone know of a good freepbx mailing list? or
can i use this mailing list for freepbx questions?
mike
On Sep 28, 2008, at 12:00 AM, Babcock, Michael Alex wrote:
thanks;
i'm messing with freepbx and what not for the time being. and am
going to give that
M, Babcock, Michael Alex <[EMAIL PROTECTED]
> wrote:
can a2 billing work on the same system that directadmin is installed?
should not be a problem
ram
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AstriCon 2008 - S
or what about if you took credit card, and billing zip code would
there be any processors that would let you do that, maybe the 3-4
digit security code on the back of the card?
mike with just some ideas.
On Sep 27, 2008, at 4:14 PM, Ruddy Gbaguidi wrote:
> Yes, we can do that. But :
> 1. we a
that would work for what i might use it for but not required right
now...
On Sep 27, 2008, at 3:16 PM, Chris Bagnall wrote:
> Most credit card processing gateways require you to have the user's
> name and address for AVS verification when you perform customer not
> present transactions. Easy
I might want something like this to, hmm.
On Sep 27, 2008, at 2:52 PM, Ruddy Gbaguidi wrote:
> Hi Guys
> We have a service that can be use by our customer via a website and
> also
> via telephone.
> On the website, we already accept credit card by sending users to
> paypal
> website where we
can a2 billing work on the same system that directadmin is installed?
On Sep 27, 2008, at 2:21 PM, broadband Voice wrote:
You need a billing software or calling card module with an IVR. You
can install A2billing in addition to Asterisk.
On Sat, Sep 27, 2008 at 6:06 PM, Babcock, Michael Alex
hi;
I'm a new member to this list and have a question for you all. I'm
sure it's something simple but alas i must ask. I've wanted to offer
calling card features to my customers. For example someone buys a
calling card from me for for say 1000 minutes, i give them a phone
number/code to cal
> Hey Sam,
>
> I've been looking for such a tool also. I can't seem to find a tool that
> does those things.
>
> If nothing comes up in the next couple of weeks I'm going to code
> something up, I wouldn't mind letting you and anyone else who might be
> i
Hi,
can you please confirm that DTMF is working properly or not?
Thanks,
Max Alex
Voip Developer
On Sat, Sep 27, 2008 at 12:24 AM, equis software <[EMAIL PROTECTED]>wrote:
> Hi, when I make a call I need that the caller can** hang up by dialing ***(H
> option in Dial command),
a
> BM 5203
> 3508 North West 114 Av.
> Doral, Florida 33178
>
> Mobile +(809)-659-0623
>
> On Fri, Sep 26, 2008 at 11:36 PM, Alex Balashov
> <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:
>
> Proxies do not handle media, so, one can definitely
new calls?
>
> --
> Dr. Haider Raza
> BM 5203
> 3508 North West 114 Av.
> Doral, Florida 33178
>
> Mobile+(809)-659-0623
>
> On Fri, Sep 26, 2008 at 10:37 PM, Alex Balashov
> <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:
>
> You
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
Mobile : (+1) (706) 338-8599
___
-- B
river to run the network interface.
> At least Dell doesn't seem to play nice with Debian.
I have not had this problem with Dell PowerEdge 2650s & 2850s but I
cannot speak to the 1950.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
D
en2&SIP/exten3,...)).
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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n.net
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
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gh to hear, depending on the size of the office).
Cheers,
AR
--
--
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[EMAIL PROTECTED]
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Register Now:
high-level application. If you want to translate various SIP
states Asterisk puts out into customised responses, you will probably
need to use something like Kamailio / OpenSIPS (OpenSER) outboard with
Asterisk to perform those translations.
-- Alex
--
Alex Balashov
Evariste Systems
Web: h
Framer: DS21552, Revision: 3 (T1)
Found a Wildcard: Digium Wildcard T100P T1/PRI
Has anyone run into this? Does anyone know what the deal is with that?
I'd dearly prefer to use 1.6.
Cheers,
-- Alex
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678
how do I get it show the codec when I'm not at
> the CLI?
Show it where, if not the CLI?
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
_
ember 22 - 25 Phoenix, Arizona
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Web
ts the problem we try to solve :)
>
> Since we have our own CDR module, we can avoid external process. What
> are the evens to listen for?
>
> Other ideas will also be appreciated.
>
> On Thu, Sep 18, 2008 at 8:23 PM, Alex Balashov
> <[EMAIL PROTECTED]> wrote:
>>
nterval.
But you are essentially correct. Things are, of course, far easier if
you just don't allow multiple simultaneous calls. :-)
-- Alex
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile :
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e in asterisk. If yes, how can i
> do it?
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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Regards
> Rizwan Hisham
>
>
>
>
> ___
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perience.
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Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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hen
> somebody adds a field to a database table whose name happens to coincide
> with an existing variable.
Or when the domain of keys you want to use for suitable or desirable
variable names does not make. :)
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel:
to use (i.e. IPSec pass-through in NAT gateways). It
can also be very difficult to get such VPNs up and running quickly
without spending money and figuring out a whole bunch of low-level
details (at least, if you really want to understand how it works).
OpenVPN's a snap.
-- Al
The OP asked, if I recall, about the protocol which is likely to be
supported rather universally by softphones and a wide variety of clients.
That is not a feature of IAX.
Tilghman Lesher wrote:
> On Friday 12 September 2008 18:31:23 Alex Balashov wrote:
>> The short answ
The short answer is SIP.
Stefan Gofferje wrote:
> http://www.voip-info.org/wiki-IAX
> http://www.voip-info.org/wiki-IAX+versus+SIP
> http://www.voip-info.org/wiki/view/Asterisk+IAX+clients
>
> Terve,
> Stefan
>
--
Alex Balashov
Evariste Systems
Web: http://www
hat one endpoint is behind
NAT on a private network that only Asterisk can see and the other
endpoint cannot. But if that's taken care of, or you have a far-end NAT
traversal solution in place to go with it, then you can do media release
on Asterisk.
-- Alex
--
Alex Bal
ng list
> To UNSUBSCRIBE or update options visit:
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>
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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Hi Hiren,
Can you please confirm the php-gd is properly installed?
Thanks,
Max Alex
Voip Developer
On Tue, Sep 9, 2008 at 4:20 PM, Hiren Mistry
<[EMAIL PROTECTED]>wrote:
>
> Dear All,
>
> I have configured here Asterisk-stat (Call Detail Records)for
> CD
suddently asterisk crashes
and i can't get email notification for received faxes.
any one help me about the crashes of asterisk?
Thanks,
Max Alex
Voip Developer
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Eric "ManxPower" Wieling wrote:
> If I am not mistaken every single echo canceler out there will disable
> itself if it detects a fax tone.
>
> Echo Cancelers do not screw up faxes, people screw up faxes. 8-)
Never underestimate how ghetto an echo canceller can be.
argument.
But, you could daisy-chain macros by calling one macro from another
using the Macro() application.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706)
I mention it in case anyone sees issue with
> what I do in the early phases that would negatively impact where I
> ultimately want to go.
Using a proxy (OpenSER) to direct distributed architectures would be
quite sensible, and they are a critical component of
Hi,
let me know that you have configured properly in res_pgsql.conf in asterisk
with proper, and it is connected properly to database with database details.
Thanks,
Max Alex
Voip Developer
On Fri, Aug 29, 2008 at 10:26 AM, Hiren Mistry <
[EMAIL PROTECTED]> wrote:
>
> Hi ,
&g
pi-digital.com --
>
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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Alex Balashov
Evariste Syst
Hi Hiren,
Have you properly configured the zap channels in asterisk,
which device have you configured in asterisk with zaptel?
let me know the dial plan for ivr.
Thanks,
Max Alex
Voip Developer
On Thu, Aug 28, 2008 at 11:40 AM, Hiren Mistry <
[EMAIL PROTECTED]> wrote:
>
> H
al()s.
-- Alex
Jon Weisman wrote:
> I'd like to do the following can someone guide me on how to accomplish this?
>
>
> Call comes in via PRI and tries to go out via SIP if for some reason the ISP
> is down and the call can not go out i want it to fail over and send the same
&
Alex Balashov wrote:
> Venefax wrote:
>
>> I need to dial a DTMF string with the Dial function using the D(“DTMF”)
>> function. What is the character for a delay? I mean, normally in other
>> technologies we use the comma to mean “wait 200 ms “. Is there an
>> eq
how many ms will the
> system wait for each comma?
w
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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ct to the home office for voice,
> data services as VPNs are extremely problematic over satellite.
Yes, indeed. Encapsulation protocols such as IPSec/GRE won't work at
all over high RTT latency (>= 400 ms).
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Te
PHP) AGI script using the CPAN JSON
parser module would suffice.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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-- Bandwidth and Colocat
also played, and dtmf is also set
properly.
But i am not getting why the incoming call is not transfer to any other
number?
Please help for this issue!
--
Thanks,
Max Alex
Voip Developer
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-- Bandwidth and Colocation Provided by http://www.api
n for this?
--
Thanks,
Max Alex
Voip Developer
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asterisk-users mailing list
To UNSUBSCRIBE
reason is TCAP (LNP, LIDB, CNAM), don't bother using Asterisk at
all.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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-- Bandwidth and Co
essages from asterisk and some
> time after this they start to complete calls normally, I don’t know what
> can be wrong. Someone has configured asterisk to wok with this
> Softswitch? Thanks for any help!
A packet capture illustrating the problem would be of utmost utility.
--
A
r
system through some voluntary action?
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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