I don't get it, if it is a web service, why do you use sockets? Isn't it just a matter of calling the web service using curl,and then wait for the response? what am I missing?Christian SavinovichVoIP Telephony Consultant646-982-3572
Original Message
Subject: Re:
directly, but if the socket won’t connect it doesn’t really matter what higher level method is used.-JustinFrom: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich Sent: Monday, February 04, 2013 9:16 AM To: Asterisk Users
Although people complaining of spam may be valid, the one part of complaining about spam that bothers me, is that some people should look at themselves in the mirror and ask the question aloud "Why does it bothers me to see another Asterisk professional compete with me for jobs?", "Why do I get
ary 10, 2013 5:17 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
On 10 Jan 2013, at 22:09, C. Savinovich c.savinov...@itntelecom.com wrote:
Unfortunately, there is a fine line between being a forum where people can exchange ideas, and being a forum w
What in the world "Asterisk to a mobile operator" means? you mean you are are using a gsm gateway? what interface are you using?... not that I intent to answer your question, but you should be clear and specific if you expect someone to give you a pointer.Christian SavinovichVoIP Telephony
the mobile operator as any other company receives calls by pots lines as T1 E1... ina ny way if he will receive calls through a gsm gateway the gateway itself must connect to pbx in a standard way probabilly voip the hardware will be a server an the interface .. hth Adriano Il 09/01/2013 18:38, C. Sav
I use the ChannelRedirect function to redirect the desired channel to the meetme roomChristian SavinovichVoIP Telephony Consultant646-982-3572
Original Message
Subject: Re: [asterisk-users] Impromptu conferencing
From: James Sharp ja...@fivecats.org
Date: Wed, November 07,
Usually, channels 1-15 and 17-31 are B-channels and 16 is the D-channel;We don't use E1s here in the USA.I just finished installing a PRI line, and being a complete novice at it myself, this is what I wish someone had told me:- the dahdi program dahdi_genconf creates 2 files 1)
, October 10, 2012 2:27 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
On Wed, 10 Oct 2012, C. Savinovich wrote:
- Not written anywhere, but the way chan_dahdi.conf is read, there are
no [] separators, so any parameters in the file apply
Hello David,If you're going to top-post (which is against the rules on this list),I apologize if mistaken, but out of curiosity, can you please refer me to where in the rules it says that we can not top-post in this list?ThanksChristian SavinovichVoIP Telephony Consultant646-982-3572
inserted in
mysql
From: Chad Wallace cwall...@lodgingcompany.com
Date: Wed, September 19, 2012 5:14 pm
To: asterisk-users@lists.digium.com
On Wed, 19 Sep 2012 13:44:47 -0700
"C. Savinovich" c.savinov...@itntelecom.com wrote:
Hello David,
If you're going to top-post (which is against
Not to bash on the developer who did this I get that we don't always think out side the box all the timeYou can bash others all you want for not thinking outside the box, but where is your effort to think outside the box yourself?. All you have to do, (that's what I did, and took me like 4 hours)
realtime don't support 'n' as
extension's next priority
From: A J Stiles asterisk_l...@earthshod.co.uk
Date: Fri, August 03, 2012 11:45 am
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
asterisk-users@lists.digium.com
On Friday 03 August 2012, C. Savinovich wrote:
N
t 03, 2012 2:21 pm
To: asterisk-users@lists.digium.com
On Friday 03 Aug 2012, C. Savinovich wrote:
You don't use 'n's in your dialplan?, you number it yourself?
man, what if you have a 300 line dialplan and then you decide to
insert a new line in the middle?
If you ever used BASIC you'
Thanks for this - but I am looking really for a software type solution.I would venture say that he means he wants it for free.Christian SavinovichVoIP Telephony Consultant646-982-3572
Original Message
Subject: Re: [asterisk-users] Question for the group
From: James Wystead
If you go to a web site called google.com, and enter "asterisk version that support the ability to have the configuration in the database", and read the first search result, you will get your answer.Christian Savinovich
Original Message
Subject: [asterisk-users] Asterisk
In my professional opinion, the phrases I don't want no Bull service and I
want the cheapest service are total contradictions. Down the road something is
not going to give.
C. Savinovich
On September 29, 2011 at 10:47 AM Danny Nicholas da...@debsinc.com wrote:
This belongs
You have plenty of ways to do this. You can use the room number + user number
to get the conference number. You can use the channel ids to keep a table of
conference members and their statuses.
C. Savinovich
On September 7, 2011 at 9:15 AM Danny Nicholas da...@debsinc.com wrote
Does this ConfBridge requires a hardware timing source? Will I be able to use
this on any virtual server without having the need special changes to the VM
setup?
Thanks
C. Savinovich
On April 25, 2011 at 10:27 AM David Backeberg dbackeb...@gmail.com wrote:
On Mon, Apr 25, 2011 at 9:38 AM
Quick question out of curiosity: Did you googled your problem, and read through
all the results, and made an exhaustive research on line of the error message
before you opted to post your question here?
CS
On April 18, 2011 at 10:16 AM Jonas Kellens jonas.kell...@telenet.be wrote:
On
You want both phones to ring? then why don't you just create a group so your
mobile also rings at the same time as the other extensions just don't answer
your mobile.
CS
On March 17, 2011 at 8:52 AM Eric Smith e...@fruitcom.com wrote:
Hi
I want to have some signal when a call is
You should try paying for the call, and then you will be able to get good
service
CS
On February 21, 2011 at 2:07 PM Christian christia...@runbox.com wrote:
Hi all,
Sorry for being a little off topic, but I just need some tips on some good
provider that offers free calls to the US. I have
45K ?
With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a
fast growing company, probably you can pay better salaries.
CS
On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote:
Job Description: Asterisk MySQL Support Engineer
Fast Growing
Can you point out to me the places in London that sell food at American prices?
Perhaps I get SeamlessWeb to deliver every morning from Brooklyn to London.
On December 22, 2010 at 1:24 PM Don Kelly d...@donkelly.biz wrote:
45K GBP would probably cover breakfast in South London. It's about
Wait, is 70k US for an experienced engineer supposed to be adequate?
Thank you, not only that , but also note that it would be 70K at the US dollar
exchange rate. However, because it is 45K Euros/Pounds earned and spent in UK,
for all practical purposes it is just the same as if it was 45K US
Without reading too much into your description, I can tell you that being an
inband sound, and as long as the dtmf tone is heard by everybody during the
conference, and being the ivr gateway one of the parties of the conference, I
don't see a reason why the ivr gateway wouldn't act upon hearing
How do you know a good soccer player when you see one? If you are a good
scout, just by his body language. Just by seeing him how he walks and
positions himself on a field. By the time he touches the ball, he is either
eliminated from my list of prospects or he is marked as good to be
He wrote me too. I would have helped him, but the name on the email address
threw me off.
CS
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Monday, November 09, 2009 9:56 PM
To:
Mr. aster...@opensourcesolution, if you had googled for how to know the
asterisk version you would have found the solution right away.
CS
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent:
Where is the log for the actual hang up of the call?.. can you do a sip
debug?
Although there can be many reasons, my first suspect is always a nat issue,
which manifest as the inability of asterisk to receive the incoming packets.
In that case, you should be getting a message saying hanging
non-zero on
'SIP/213.165.32.100-b7c10ad8' in macro 'hangupcall'
== Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on
'SIP/213.165.32.100-b7c10ad8'
elastix*CLI
thanks
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C
What about if I use the browser from my cellular phone?
CS
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno
Sent: Wednesday, September 16, 2009 10:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
It all depends what are you going to use Asterisk for. Sounds like it is
for conferencing. Would you care to elaborate?
CS
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mauro Sergio
Ferreira Brasil
Sent:
Let me see if I get you: you inserted the installation CD, then you
restarted the computer, and now you want to know what to do next?
CS
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Poe
Sent: Friday,
Very few calls have been made this way, trivial cost, but it is slightly
worrying.
That's what I thought when they hacked into one of my systems, but it is
not the cost of the calls, it is the purposed of the calls you should watch
out for. The FBI contacted the owner of the PBX, and inquired
Nothing is difficult my friend. If you dedicate a few cups of coffee to
it, a couple of days, and do some good googling, you will get it done
yourself.
Good luck!
CS
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
As far as I know, just reinstall 1.4.20, and your problem goes away.
CS
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vikas
Sent: Wednesday, April 22, 2009 4:12 AM
To: asterisk-users@lists.digium.com;
I am having a similar issue. Asterisk does not show ringback tone and I
investigated this due to it not reading sip invite 180. (or supposedly not
receiving it).. My solution is that now I am using h323
(ver 1.4.19)
CS
From: asterisk-users-boun...@lists.digium.com
Alex, tu hablas español?
CS
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Sunday, April 12, 2009 9:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Dear Sir:
Me no peak englisss…. 20 pesos??? ok , tank you sir, tank you sir
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kinjal Dixit
Sent: Tuesday, March 03, 2009 9:57 PM
To: Asterisk Users Mailing List -
Asterisk compatible One to many video is achieved with VidPhone. You can
download the web embedded video component free by signing up an account on
my website www.itntelecom.com. Any help on usage, just send me a note and I
will be glad to help you set it up.
CS
-Original Message-
I recently completed PhoneClient 1.2 which is a Windows executable that
interfaces with Asterisk, with capacity to receive numbers from the
clipboard via a hotkey. PhoneClient is also a call manager and meeting room
interface. If interested please contact me off the list.
C. Savinovich
directly.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich
Sent: Friday, January 30, 2009 12:03 PM
Type how to build an asterisk PBX in google
CS
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of nightduke
Sent: Sunday, January 25, 2009 9:09 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
None of these examples actually create a 3-way call, which is, unless I am
mistaken, the original request. An incoming/outgoing call gets bridged to a
local channel alright, but then how do you bridge that call to yet another
call?.
I did try some alternatives and the only way I found is by
Your asterisk is using 99.9% of cpu and you want it to use more? Do you
mean you want asterisk to use LESS cpu?
CS
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Kim
Sent: Friday, December 19, 2008
Greg's question is this:
- Does anybody has a sample on how to open and query a Microsoft SQL
database from the dialplan?(and which are the correct drivers/addons to
install?)
Thanks
CS
-Original Message-
From: asterisk-users-boun...@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Text messaging and Asterisk
C. Savinovich wrote:
Can somebody please give a pointer to a complete neophyte (like me) on
text messaging, what product can I use to send and automatic text message
to
a cell phone from
Can somebody please give a pointer to a complete neophyte (like me) on
text messaging, what product can I use to send and automatic text message to
a cell phone from within the asterisk dialplan? (the part of the dialplan I
have down, the part of the text message no)
Thanks
C. Savinovich
-Commercial Discussion
Subject: Re: [asterisk-users] Text messaging and Asterisk
C. Savinovich wrote:
Can somebody please give a pointer to a complete neophyte (like me) on
text messaging, what product can I use to send and automatic text message
to
a cell phone from within the asterisk dialplan
] Text messaging and Asterisk
On Oct 13, 2008, at 11:28 AM, C. Savinovich wrote:
I mean is if someone know of an sms server or service that allows
me to
send outgoing text messaging.
Are you sending SMS to known users or to any mobile phone user?
If you are sending to a fixed user base
I am tinkering with a new router, a Linksys wrt610n dual-band, etc. But
the when I connect it, the softphones(x-lite) on the computers don't even
register. After a couple of hours of hassle, I found out that if I dmz the
router to the computer I am using, the softphone starts to work.
I don't see where it is difficult to figure out.
First of all, system keeps looking up on the table as user dial each
number.
When number starts with 1, expect USA. When number doesn't start with
either 1 nor 0, expect USA too.
When number starts with 011, and as country code and city
It's amazing... the man starts the thread with a simple question: Can
anybody tell him if Asterisk can do the same things that the Cisco Unity
Server can do?, if it can do some better, some the same, and/or some worse,
can someone indicate which ones? Also, can Asterisk complement the Cisco
I am puzzled by the quality of magicjack. I keep trying to figure out how
they can the quality be that adequate. Since Skype also has an excellent
quality, that leaves me to believe that software based calls (softphones)
could have and advantage over hardphones, provided there is a parameter
PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MagicJack quality
On Fri, 11 Jul 2008 17:13:15 -0400, C. Savinovich wrote:
I am puzzled by the quality of magicjack. I keep trying to figure out how
they can the quality be that adequate. Since Skype
Yes, I have designed two different webphones, granted, using third party
libraries, and magicjack's quality is better. I acknowledge that.
Thank you, but referring me to someone's review won't help me much... I am
interested in the internals. Regardless, their technique has a twist, and
or disable them.
I might buy one just to hack it. Has anyone sniffed it or poked
around at all on lists?
Thanks,
Steve T
On Fri, Jul 11, 2008 at 6:03 PM, C. Savinovich
[EMAIL PROTECTED] wrote:
Yes, I have designed two different webphones, granted, using third party
libraries, and magicjack's
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Centile ipbx, anyone heard of this?
On Tue, Jun 24, 2008 at 5:58 AM, C. Savinovich
[EMAIL PROTECTED] wrote:
To be fair, Centile is better geared than asterisk for virtual pbx
hosting. It comes with a system to manage virtual pbxs
I do. I spent about a month doing contract work for a company that
provides pbx hosting. Their hosting is based on Centile. What do you want
to know?... they had about 80 customers (virtual pbxs), 320 ip phones... it
seemed to be running ok. I could have done the same thing with asterisk :)
Discussion
Subject: Re: [asterisk-users] Centile ipbx, anyone heard of this?
On Mon, Jun 23, 2008 at 10:01 PM, C. Savinovich
[EMAIL PROTECTED] wrote:
seemed to be running ok. I could have done the same thing with asterisk
:)
Basically, I was really curious why a company would use this instead
Check the web embedded click-to-call solution from videoreps.net. It is
free. It includes click-to-video, click-to-call, and click-to-did
CS
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of equis software
Sent: Wednesday, April 16, 2008 7:36 AM
To: Asterisk Users
No problem. The program is in Windows. Contact me off line to make
arrangements to send you the installation files.
C. Savinovich
Long ago, I wrote a nice program that reads CDR output from any
legacy PBX via the serial port. Not much in use lately, but I will be
happy to furbish
asks.
C. Savinovich
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You can try the click-to-call from www.videoreps.net... it is asterisk
based. The sample provides you with an actual pc-to-pstn call... of
course calls to internal extensions are easier.
There is click-to-call, and there is click-to-call-with-video
CS
somebody knows some application web
Yes... and there is plenty of information about sip-to-sip communications
if you do research
CS
i want to use Cisco AS5400 media Gateway as my PSTN Gateway instead of
using the E1 PCI cards in asterisk box ,is this practically
possible? can i use SIP in the connection between Asterisk and
It is doable. The iPhone uses a subset of the Apple OS. Sometime ago I
reviewed the file structure of the iPhone. It is just a matter of placing
the voicemail files from * into the voicemail folder of the iPhone.
Somebody with more time than me though :)
CS
-Original Message-
Dear All:
Just as the name suggests, and evolving from regular Click-to-Call,
Click-to-Call WITH VIDEO provides web sites with the ability to engage
their visitors with a live video agent (plus the phone call). All with just
a click of a button placed on the customer's web site. Please
OK gentlemen, thank you very much.
Best Regards
C. Savinovich
VideoReps.net
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: Thursday, September 20, 2007 11:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
welcomed to use
videoreps for free until you establish how it can benefit you.
C. Savinovich
VideoReps
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Thursday, September 20, 2007 11:24 AM
To: Asterisk Users Mailing List - Non
it.
C. Savinovich wrote:
Please don't change the title of my post. It is disrespectful. One thing
is to give your opinion about its content, and another to be self
appointed
editor of this forum.
If you agree or disagree with it fine, but let
others decide. They know spam when
Can people on this list share their experiences on how they partition a DSL
for small business internet service with a router so that a portion is
dedicated to VOIP and another portion to computers. Of course, the idea is
to do this with a low cost router (under $100).
Many Thanks
C
Looks good. a lot of initial work, but looks worth the effort. Do you find
that it improves the quality of your VOIP calls?
C. Savinovich
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Robar
Sent: Monday, September 10, 2007 11:28 AM
To: Asterisk Users Mailing List
I don't have much details on your set-up, but I assume that since
quintums had performance troubles with SIP (about 2 years ago) your best
bet is to get them to work with h323. For that your first step willl be
to install h323 support on your asterisk box. I may be a little rusty
on this, so
Hello everyone, I am pulling my hair here because a carrier threw me curve
early today.
They want to send calls to my asterisk server using SIP. Then they said
that their gateways don't have to register with my server, that all they have
to do is send a prefix for validation. Whereas
Randy:
Yes, you are right, thank you and consider this the end of this
'unfortunate' thread :) . See, I acknowledge it was wrong to send the
test message, but that doesn't mean people have to be rude in pointing it
out.
Bye everyone
CS
-Original Message-
From: Randy
This is a test,
please disregard
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On Tue, 2005-01-11 at 18:30 -0500, C. Savinovich wrote:
This is a test, please disregard
Steven Critchfield [EMAIL PROTECTED]
Next time you post, make sure you turn off HTML.
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http
: Tuesday, January 11, 2005 5:44 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] test-ignore
On Tue, 2005-01-11 at 18:30 -0500, C. Savinovich wrote:
This is a test, please disregard
Next time you post, make sure you turn off HTML
You know, you have a choice.
C. Savinovich
ITN-Telecom
I don't know. Dealing with some people here seems like I am in hell
tilting at dragons.
On Tue, 2005-01-11 at 20:02 -0500, C. Savinovich wrote:
Brian:
Do you mean Don Quixote and the windmill?
CS
Dante
I don't mean to be patronizing at all, but one thing I've learned is that
if a customer does not want to spend $150 on anything, either he/she is an
incredibly shrewd businessman (because he is going to make you bust your
balls for free until he sees results), or he is really not interested and
I am trying to set
up XPro behind a Squid Proxy. What should I put in outbound proxy?, what
is a STUN server?
Thanks
C.
Savinovich
ITN-Telecom
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Dean:
We have a commercial videoconference product. The closest we can get
is to initiate the VC based on the phone call started by asterisk, which
would be really cool, but there will be a charge for the video software.
C. Savinovich
ITN-Telecom
212-865-9118
-Original Message
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