it to a restricted range of IP addresses?
Nicholas Blasgen
Partner / Network Operations
Refractive Dialer LLC
(724) 252-7436
On Sun, Jan 3, 2010 at 8:29 PM, Max McGraw max.mcg...@gmail.com wrote:
Nicholas,
you haven't specified which version, which does make
a lot of difference.
1.6.x can
assumed none, but I can't even get replies on registration from any of my 3
VoIP providers. I tried defining the External IP and some other stuff, but
I assume it's fully an issue with the firewall. Do I really need 5060 port
forwarded just to register with remote hosts?
Nicholas Blasgen
Partner
the business version of Skype.
Nicholas Blasgen
Partner / Network Operations
Refractive Dialer LLC
(724) 252-7436
On Tue, Aug 18, 2009 at 10:35 AM, Pascal Bruno tipas...@gmail.com wrote:
Lol but he has a good point and makes a lot of sense. Never thought about
that strategy...
On Tue, Aug
I'm sure I saw a MS C++ library that had additional support to be wrapped up
as an ActiveX client. But I can't seem to find anything now. SIP ActiveX
clients are around.
Or maybe this is it:
http://www.secondsignal.com/secondsignal/sshome.nsf/html/2ndSignal-IAXClientWrapper2005
Nicholas
if the Dial'ed line
hangs up it returns back to the orginal Dial Plan. Doesn't help at all.
You hang up on the person, the person goes to the next line in the dial
plan, and you get called again. You hang up, they call you back again.
Soulds like a good way to use up air time.
Nicholas Blasgen
Partner
available.
If you don't get any help, you can try opening it as a bug on Digium's Bug
Tracker but I assume the issue isn't a bug but just an overloaded system
with a slow response time.
Nicholas Blasgen
Partner / Network Operations
Refractive Dialer LLC
415.692-5277 (w)
408.497.9796 (c)
Please update
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate
ActionId and Account can be set.
Nicholas Blasgen
Partner / Network Operations
Refractive Dialer LLC
415.692-5277 (w)
408.497.9796 (c)
Please update your contact records with my new work number.
On Wed, May 27, 2009
/05/2009 3:44 p.m., Nicholas Blasgen wrote:
Has anyone else had issues with Originate returning the wrong error
code? According to the docs, the following errors are supposed to be
returned:
0 = no such extension or number
1 = no answer
4 = answered
8 = congested or not available
AM, Nicholas Blasgen
nicho...@refractivedialer.com wrote:
Matt,
Oh, I thought it was Asterisk 1.4.23 like I wrote in my first email, but
turns out to be Asterisk SVN-branch-1.4-r191778.
But yes, I am talking about originateresponse. I'm going to do some more
debugging today to see if I can
).
Any ideas to correct this issue? Or is there a better updated version of
that list that would fix my understanding of what the error codes were?
Nicholas Blasgen
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asterisk-users
Thank you Mark. I did try it out myself and figured out that it did work as
I wanted. Thanks for the quick reply though.
Nicholas Blasgen
Partner / Network Operations
Refractive Dialer LLC
408.395.2110 (w)
408.497.9796 (c)
On Mon, Feb 2, 2009 at 12:06 PM, Mark Michelson mmichel
that it shouldn't work.
So the question is, how can I listen into a channel if I know either the
channel or the unqiue id? And in the meantime I will play around with
ChanSpy more.
Nicholas Blasgen
Partner / Network Operations
Refractive Dialer LLC
408.395.2110 (w)
408.497.9796 (c
Interface (AMI) to perform a Redirect on the person
you're talking to. Doing this causes the AGI script to continue.
--
Nicholas Blasgen
[EMAIL PROTECTED]
408.497.9796 (c)
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--
Nicholas Blasgen
[EMAIL PROTECTED]
408.497.9796 (c)
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is returned
and my AGI script can continue. So I think it should be fine. Has anyone
done anything like this? Any pointers would be great.
--
Nicholas Blasgen
[EMAIL PROTECTED]
408.497.9796 (c)
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-users
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Nicholas Blasgen
[EMAIL PROTECTED]
408.497.9796 (c)
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asterisk-users mailing
00, len 000160)
Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160)
Asterisk SVN-branch-1.4-r118365
--
Nicholas Blasgen
[EMAIL PROTECTED]
408.497.9796 (c)
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Actually, I thought about it for a while. What I want is something that
will allow me to restart the message if another sound is detected.
Something like this:
exten = answermachine,1,Answer()
exten = answermachine,n,WaitForSilence(1000,2)
exten = answermachine,n,Background(message)
exten =
, but they shouldn't
be needed. Anyone have a suggestion?
--
Nicholas Blasgen
[EMAIL PROTECTED]
408.497.9796 (c)
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Is there any way to see the number of AGI processes that Asterisk is
handling? Either console, Asterisk Manager, or from within the AGI? I used
to just count the number of running copies of my AGI process (ps aux | grep
agi) but once in a blue moon one of my AGI processes will become a zombie or
=0057510
secret=0057510
fromdomain=directnationalloan.com
outboundproxy=las-obproxy.voipzone.us
host=directnationalloan.com
insecure=port,invite
qualify=yes
type=peer
On 1/17/08, Nicholas Blasgen [EMAIL PROTECTED] wrote:
I've set up plenty of Asterisk boxes but never one that had
I've set up plenty of Asterisk boxes but never one that had to deal with a
proxy server to be able to use a line. Using X-Lite I have no issue with
settings as follows:
Display Name: Any Name
User name: 0057510
Password: 0057510
Authorization user name: blank
Domain:
I'm no longer on the DEV mailing list, but:
# svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk
svn: URL 'http://svn.digium.com/svn/asterisk/branches/1.4' doesn't exist
http://svn.digium.com/svn/asterisk/branches/
--
/Nick
___
Some email asked for some examples. He's an example system that
will use ViaTalk lines (which allow 2 concurrent calls on a channel,
so I use GroupCount to check for a value of 2). It isn't round-robin
and actually I'd pay someone good money to make a revised Dial()
function that would do round
I don't understand the
USERS vs PEER vs FRIENDS. I just use Peer for everything. Has to do
with can I only contact you or can you contact me too? ... Peer does
it all.
RealTime does have an issue. If you don't turn on caching, then it holds no
state information. So if you think you're going
I've got a very nice PHP AGI script but I want to be able to do some
database cleanup when the user hangs up the phone. I wish everyone would
hang up when they were suposed to, but some people don't. So what does
Asterisk send to an AGI file when the line has been disconnected? If I
remember
It gets hard to read my logs when every time someone makes a phone call it
displays long pages of Dropping voice frame. Anyone encounter this
before? Asterisk is bridging two SIP lines together, so the technology
should be the same. Maybe I'll try allowing only ULAW.
Okay, you need to explain to me a little more about why you're calling the
list before connecting it to an extension.
So for me, I use a .CALL file but I assume your setup does the same thing.
It will call a number and once ANSWERED, pass it to an extension. Let's say
we pass it to a LOCAL
Setting call-limit=1 in sip.conf will limit the number of incomming (and
outgoing?) channels on your SIP device to the number you specifiy (1 in this
case). If you want to allow more outgoing, but only 1 incomming, you could
do that with some GROUP() checking. Problem is that when there isn't an
Once upon a time it cost $20/hr over a 9600 baud link to read stuff
like this and people tended to think before they asked questions,
I'm afraid they didn't, I vividly remember asking inane questions at
1200baud
over uucp ;-)
Ditto. I still have newsgroup posts of mine asking about
It would surprise me to see mySQL configured incorrectly, but it's always a
possibility. Look at the mysql server var called 'wait_timeout'.
phpMyAdmin shows it under system vars.
I've got a macro that tries to find the first available SIP trunk to send
outgoing calls on. It tracks the usage of the lines (since each trunk has a
call-limit of 2) by using GROUP(). My problem is that once a call switched
to ANSWER state, ``group show channels`` stops listing it and then my
exten = 555,1,Dial(Local/1234567890/n)
note the /n
I'm going to try this in a bit (can't hurt anything, might as well), but I'd
like to understand you're reasoning. You're dialing an extra extension?
I'm also going to be trying this with Asterisk 1.6 TRUNK to see if it's even
a current
Just thinking about it quickly, it's always possible it has nothing to do
with Asterisk. There are many instances where I run into issues with a
poorly configured servers when they have even a little bump in HTTP
traffic. This was years ago though, and it was an issue to do with a web
server and
I've heard about this, but I really can't seem to find anything on it. I've
got a strange setup that exists only because of firewall issues, and
everything about it seems fine. The setup:
SIP clients - Asterisk (office) - IAX - Asterisk (colocation) - SIP PSTN
Termination
All the extensions I
At least with my Manager API, I have the ability to simply set a default
event handler and using that I can dump all events as the pass though. Then
I setup a case switch and act on the ones I want.
But the manager events I like are LINKED and HANGUP.
Ah, I correct myself. I see, you wanted to know the headers for each SIP
packet. Makes a lot more sense now.
On 8/15/07, Anthony Francis [EMAIL PROTECTED] wrote:
http://www.faqs.org/rfcs/rfc3261.html
Rizwan Hisham wrote:
Hi All,
Can anybody send me a complete list of sip events. i know
So besides the missing ) on line 1, I have some other comments:
1) You should replace your priority numbers with 'n'. Just so much easier
to know that the issue isn't with priority numbers. And typing 'dialplan
show context' is a nice way to see if everything is setup correctly. The
'n' is a
I have 10 SIP trunks that I'd really like to round-robin load balance.
Currently I have a macro that switches between available lines, but there
really must be a function in Asterisk to do this on its own. So my question
is just that, are there any easy ways for Asterisk to either balance between
First, it seems I have to have a 2 - 3 second wait before the AGI call in
order to get valid CID data. Usually 2 seconds suffices for this one
setup
but during that time the caller has had two rings before the local
extension
has even begun to ring. Is there something I am doing wrong
I've got 4 SIP phone lines with a call-limit of 2 for each. I've written a
handy macro to allow my users to dial a phone number and the macro will
figure out the next available line to use by first checking if the GROUP()
is over 2 and then checking to see if ChanIsAvail() as a backup, and if it
] wrote:
On Tue, 2007-08-07 at 11:13 -0700, Nicholas Blasgen wrote:
My question is this. Is it possible to tell Asterisk to execute part
of a macro as a block without allowing any other commands to be
processed during that time?
You'll want to check out the MacroExclusive() application
Not specific to the SPA3102, but just normal outbound dialing is as follows:
exten = _1NXXNXX,1,Dial(trunk type/name/${EXTEN})
or if you want to require people to dial 9, then:
exten = _91NXXNXX,1,Dial(trunk type/name/${EXTEN})
or if you're like me and you're used to a cell phone and
I'm running Asterisk without FreePBX or any of the other managers. I'm
trying to figure out how to set the maximum number of channels allowed on a
single line? I'd just rather not have Asterisk try the line when I know
I'll recieve a CONGESTION back from the SIP phone provider (ViaTalk in this
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