For what it's worth, I see similar traffic regularly from:
orange.ps
hadara.ps
ovh.net
iweb.ca
scalabledns.com
securedservers.com
wholesaleinternet.com
hostnoc.net
rackspace.com
hetzner.de
all going to 972-59-* numbers (i.e. Paltel/Jawal mobile customers).
Common numbers are:
972592871970
On Aug 6, 2013, at 2:59 PM, Chris Bagnall aster...@lists.minotaur.cc wrote:
FWIW, we routinely see dodgy traffic from:
ovh.net
hetzner.de
But since those are 2 of the larger short-term contract dedicated server
vendors, I'm not surprised about that. It's so frequent that I don't even
I'm using Asterisk 1.8.6.0 on my router talking to my ISP's Taqua 7000 (?)
switch.
I'm using a config that looks like:
[sip_proxy-out]
type=peer
authuser=208
remotesecret=xyzzy
qualify=100
host=n.n.n.n
call-limit=5
nat=no
; sendrpid=yes
insecure=no
But the Taqua responds to outbound
Anyone have any configuration experience connecting Asterisk 1.8 to the PSTN
via SIP on a Taqua 7000 switch?
My local carrier recently upgraded software and changed their configs so that
signalling and media are on different cards (and hence different IP addresses),
and it's causing issues.
I
Anyone know where the sources for app_nv_faxdetect officially live? I
couldn't turn them up on a web search, just patched versions for 1.4, etc.
Thanks.
--
_
-- Bandwidth and Colocation Provided by
On 7/25/10 7:54 PM, Nick Brown wrote:
Hi All,
Facing an issue at the moment with setting the TOS on packets - the
documentation is a bit light, however is straightforward so unsure if this is
a configuration issue or a bug.
Following is set in sip.conf;
tos_sip=CS3
tos_audio=EF
And
On 7/24/10 1:23 PM, Ira wrote:
At 10:01 AM 7/24/2010, you wrote:
The version of gcc is most definitely related to this problem, although it is
the linker that is the problem. My recommendation for solving this problem
is to use 'make menuselect' to eliminate any modules that you do not wish
Sounds like a great ear warmer!!!
Hell, you can probably grill a panini with it if you're patient.
On 7/23/10 6:39 AM, Matt wrote:
You're using phones that draw 15Watts?!?! Let me know what brand this is so I
can stay away from them.
On Thu, Jul 22, 2010 at 2:58 PM, David Gibbons
On 7/23/10 6:18 PM, Ira wrote:
At 02:58 PM 7/23/2010, you wrote:
The Asterisk Development Team has announced the release of Asterisk
1.8.0-beta1.
So being the brave type, I downloaded and installed this onto my
Asterisk Box. Compiled fine and installed fine, but it didn't work.
I kept
a.
i On 5/22/2010 at 3:28 PM, Philip Prindeville
philipp_s...@redfish-solutions.com wrote:
For the 3rd consecutive term, the US Senate has introduced the Truth in
caller ID Act of 2009.
It was passed by the Senate (finally) in January, and has moved to the
House for a vote.
A lot
For the 3rd consecutive term, the US Senate has introduced the Truth in
caller ID Act of 2009.
It was passed by the Senate (finally) in January, and has moved to the
House for a vote.
A lot of states have ambiguous or overly restrictive language on how
caller ID may be manipulated.
For
Anyone have any luck configuring a SIP trunk on a Taqua to talk to Asterisk?
We were initially set up as a subscriber (access line) but that had some
undesirable side-effects, such as quashing the ANI on outbound calls.
Looks like we're going to have to reconfigure the trunk as a network
On 4/29/10 4:22 AM, Jim Dickenson wrote:
I banged my head with a like problem a few days ago.
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
n does not mean the letter n in a pattern it has a special meaning!
That's capital N, isn't it?
Also, the prefix _stdexten-. seems to
On 4/29/10 1:55 PM, Tilghman Lesher wrote:
On Thursday 29 April 2010 13:11:17 Philip A. Prindeville wrote:
Doesn't quite make it 'deterministic' if you have to test it to see what
it's going to do.
The code is deterministic. The human who wrote the example is not. Are
you
Just saw from build 2036:
Starting mini_httpd...
WARNING WARNING WARNING
YOU STILL HAVE NOT CHANGED YOUR HTTPS ADMIN PASSWORD
ANYONE THAT KNOWS YOU ARE USING ASTLINUX CAN DESTROY YOUR
SYSTEM. PLEASE CHANGE THIS OR DISABLE THE HTTPS ADMIN
INTERFACE IMMEDIATELY!
Example:
htpasswd
Darrick Hartman wrote:
Philip Prindeville wrote:
Just saw from build 2036:
Now, to get the following packages to build:
misdn
asterisk-chanmisdn
nistnet
rhino
strace
rp-pppoe
Whoops. I'm sure Philip thought he was sending this to a different
mailing list
Andres wrote:
I'll look into using Record() or Monitor() to capture the phone call,
but if there's any conversion being done by codecs then that won't
eliminate the possibility that the code itself is misconfigured or buggy
and generating a bad stream on one of the legs...
Anyone have an
Philip Prindeville wrote:
Well, things just got a lot more interesting... Adding Monitor() to an
extension ends the one-way voice problem on inbound calls!
So an incoming call gets handled as:
[ctc-incoming]
exten = 208345,1,Noop()
exten = 208345,n,Log(NOTICE: RDNIS: ${CALLERID
? I'm baffled (no pun intended). And is there any debugging I
can turn on to reveal CODEC behavior that might differ from 113 and 119?
Thanks,
-Philip
Philip Prindeville wrote:
I've got the following situation. I'm running Asterisk 1.4.18 on a
firewall/gateway machine, with some SPA-942 (f/w
I've got the following situation. I'm running Asterisk 1.4.18 on a
firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones
behind it.
I'm peering SIP with a Coppercom switch sitting behind an SBC.
On outbound calls, I get 2-way voice, no worries.
On inbound calls, I get one-way
I've been working on getting the sample configuration of extensions.conf
to be more usable, i.e. to make the examples be more flexible or cover
more territory... I thought it might be handy to show people how to use
more contexts for virtual hosting, for example.
Problem is I was using the
Sorry for the late follow-up to this... it was on my to-do list for
over a month... Sigh.
I've submitted a configuration bug and for this:
http://bugs.digium.com/view.php?id=11969
The hope being that if the examples provided in the configs/ directory
work better out of the box for
I'm currently getting SIP trunking from my PSTN provider, but they don't
quite grok the whole any-service/any-device philosophy...
I'm wondering if it's possible to get SIP voice carriage from one
provider, but have SMS associated with the same phone numbers being
provided by another carrier?
I'm looking to ditch my GSM phone with Cingular and provide my own
calling services with Asterisk hanging my phone off the Edge or 3GPP
network... Luckily, I have the option (through my employer) of getting
a data-only plan.
My question is, other than the Nokia E61i or E70, what phones will
Luki wrote:
I always assumed that you can have multiple SIP phones behind a Linksys
firewall/router (WRT54G) all using the same STUN server/port.
I got 10-20 SPA942's behind a OpenWRT router (on WRT54G, WRTSL54GS,
...) at several sites, no STUN, no special configuration, no problems
at
Howdy,
Excuse the neophyte questions... I was wondering:
(1) what's involved in setting up a call with encrypted media (I'm on a
cable network and don't want my calls snooped);
(2) is there a cheat-sheet for configuring Sipura handsets/hardphones
like the SPA-942, and in particular for
Uhhh... just
export HOSTNAME
should be enough once it's been set.
Joost Kuif | Mobillion wrote:
This pointed me into the right direction, thanks Tzafrir!
i added a export HOSTNAME=$HOSTNAME into my .bash_profile
Grtz,
Joost
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
Chris Bagnall wrote:
(2) is there a cheat-sheet for configuring Sipura handsets/hardphones
like the SPA-942, and in particular for message-waiting indicator and
shared-line appearances?
MWI is easy... simply add a [EMAIL PROTECTED] setting to sip.conf
for the phone
Make sure
I was wondering under what conditions Asterisk will hand off a call to
another switch.
I'm trying to verify that my local PSTN's Coppercom switch operates
correctly... and wanted to know how to get a call REFER'd to another
end-point.
Thanks,
-Philip
In trying to get my services up and running, I've encountered the usual
spate of first-time issues.
I was wondering if there was a good FAQ or Howto on troubleshooting NAT
issues.
The equipment that I'm using is typically a Sipura ATA or hardphone
(SPA-942) sitting behind either a Linux box
Lee Jenkins wrote:
Vincent wrote:
On Fri, 14 Dec 2007 10:51:10 -0500, Lee Jenkins [EMAIL PROTECTED]
wrote:
I have to reboot my desktop xp box daily for it to run well.
I haven't rebooted my XPSP2 in months, and I let it run 24/7, with a
bunch of apps open at all times. And
Philipp Kempgen wrote:
Anselm Martin Hoffmeister wrote:
In most cases it seems to end at
the fact that providers correct caller-ids they get from the calling
party: If you send any number which is assigned to the PRI (or SIP
trunk), that is fine; if you send another number, it will be
I've got the following set up:
Someone calls into my PBX on a single number (via SIP trunk from my
carrier), and the get a voice menu of extensions.
On one of the extensions, it rings a bunch of internal SIP hardphones,
plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN
Doug wrote:
At 19:55 12/13/2007, Vincent wrote:
Hello
I was wondering why there doesn't seem to a Windows version of Zaptel,
making the Digium and its clones unavailable for a Windows PBX.
Is the Zaptel/Zapata combo too *nix-centric?
Thanks.
Windows is a half-baked, dying OS
Tilghman Lesher wrote:
On Thursday 13 December 2007 19:55:39 Vincent wrote:
I was wondering why there doesn't seem to a Windows version of Zaptel,
making the Digium and its clones unavailable for a Windows PBX.
Because nobody has done it yet. The real answer is probably more along
Ok, I think I asked this previously but don't remember seeing an answer...
Yes, you can tickle an SPA94x or 962 and have it fetch a config from a
TFTP server... But is there no way to simply push a couple of lines
of XML config to it directly via an HTTP POST (sans TFTP server)?
Thanks,
Kristian Kielhofner wrote:
On Dec 9, 2007 2:05 AM, Philip Prindeville
[EMAIL PROTECTED] wrote:
..snip..
You think that an Asterisk configuration is a lot larger than a Cisco
5850 Access Server or a 7216 core router?
IOS doesn't use XML for configuration. What's a 7216
So, for the hotel project, what was the conclusion? I don't remember
seeing a summary.
And were any of the phones combination Wifi/GSM? Like a Nokia E70 or E61i?
I've been looking for such a phone to use myself, but so far haven't
found one that I liked. Common flaws were:
* poor standby
Eric C. wrote:
Hello all,
Just starting to setup asterisk v 1.4.11 and need to run three distinct phone
systems for three different companies.
So far, I have inbound lines going to the appropriate dial plan within the
extensions.conf file. I'm using
exten = _X.,1,NoOp(FROM NUMBER:
Ryan Burke wrote:
Tilghman Lesher wrote:
On Friday 07 December 2007 20:12:12 Philip Prindeville wrote:
Darryl Dunkin wrote:
You can store most of the configurations in a database which may be
more
accessable to you.
Perl can also parse these configurations quickly
Tilghman Lesher wrote:
On Saturday 08 December 2007 00:51:44 Philip Prindeville wrote:
Tilghman Lesher wrote:
On Friday 07 December 2007 20:12:12 Philip Prindeville wrote:
Darryl Dunkin wrote:
You can store most of the configurations in a database which may
Jared Smith wrote:
On Sat, 2007-12-08 at 13:55 -0800, Philip Prindeville wrote:
Going back to my original posting, I was also suggesting that the parse
tree from Asterisk could be read in and then dumped out as XML, so that
other software could then ingest it... using it as a common
Tzafrir Cohen wrote:
[snip]
3 of the handset manufacturers that I use, 1 of the firewalls, and 2 of
the video-conference engines all use XML. And the list gets longer
every day.
Most of the programs I have don't use XML. And I only feel better.
Eventually, they will start to
I'm starting work on some provisioning tools to simplify plugging in and
configuring hard SIP handsets and conference bridges (maybe eventually
MPEG-4 PoE video cameras that speak SIP as well).
Issue is that I'd like to glean as much information out of the
configuration files... but don't
That's sort of my point: that you have to reinvent it, and it's easy to
get wrong.
Darryl Dunkin wrote:
You can store most of the configurations in a database which may be more
accessable to you.
Perl can also parse these configurations quickly enough if you know how
to use the input
Tilghman Lesher wrote:
On Friday 07 December 2007 20:12:12 Philip Prindeville wrote:
Darryl Dunkin wrote:
You can store most of the configurations in a database which may be more
accessable to you.
Perl can also parse these configurations quickly enough if you know how
to use
Bill Andersen wrote:
Philip Prindeville wrote:
So I'd venture to say that by August, the Internet will really be *30*
years old.
As Al Gore was born in 1948, I can see that the Internet could be as old
as 30, but not much more. 35 years ago would put him at 25 years old
Tilghman Lesher wrote:
On Friday 07 December 2007 09:56:56 Bill Andersen wrote:
Philip Prindeville wrote:
So I'd venture to say that by August, the Internet will really be *30*
years old.
As Al Gore was born in 1948, I can see that the Internet could be as old
as 30
The Internet is a lot older than 20 years!
I sent my first email in 1981 (yes, using SMTP and TCP/IP), and even
then it had been around for a while (maybe not using TCP... but in the
NCP incarnation at least). IP dates back to... what? 1978?
Yeah, the RFC was published in September 1981, but
need Batman, and it's a
perfectly cloudless night? Or what if it were foggy?)
Will they send me a sign if my feature gets approved? Should I look out
the window towards downtown? ;-)
Or... am I really supposed to file a bug after all?
Philip Prindeville wrote:
Hi.
I wanted to write
Joshua Colp wrote:
- Original Message -
From: Philip Prindeville
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List
- Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Thu, 06 Dec 2007 22:34:18 -0400
Subject: Re: [asterisk-users] New feature:
calling all bug
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Ryan Burke [EMAIL PROTECTED] wrote:
I just was looking over the app_waitutil.c and am confused you add 500 to
tv.tv_usec on the line msec = (future - tv.tv_sec) * 1000 - ((tv.tv_usec
+ 500) / 1000);?
It's just doing a standard
Ira wrote:
At 11:58 PM 12/4/2007, you wrote:
You used to be able to dial popcorn (767-2676) in any area code (at
least prior to 1982) and get the current time.
I thought it was UL3-2121 when I was younger and occasionally if that
was the only number in the UL3 prefix, dialing
-- An open source telephony toolkit.
*
* Copyright (C) 2007, Redfish Solutions
*
* Philip Prindeville [EMAIL PROTECTED]
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance
Steve Edwards wrote:
On Tue, 4 Dec 2007, Philip Prindeville wrote:
I wanted to write a popcorn app for myself, both to learn how to script in
Just out of curiosity, what does this have to do with popcorn?
Thanks in advance
Anthony Francis wrote:
Philip Prindeville wrote:
Tilghman Lesher wrote:
On Thursday 29 November 2007 13:29:17 Philip Prindeville wrote:
[snip]
The issue is that I have, per virtual pbx (i.e. home or business), two
contexts that these get used from
Tilghman Lesher wrote:
On Thursday 29 November 2007 13:29:17 Philip Prindeville wrote:
[snip]
The issue is that I have, per virtual pbx (i.e. home or business), two
contexts that these get used from. The internal-xyzzy and
incoming-xyzzy contexts (one for each pbx, ie. xyzzy is home
bump...
Philip Prindeville wrote:
I'm trying to set up my extensions.conf file using some of the existing
macros like stdexten, etc. while at the same time having two logically
separate virtual PBX's (with no default context) and two trunks coming
into separate contexts, i.e. one
I'm trying to set up my extensions.conf file using some of the existing
macros like stdexten, etc. while at the same time having two logically
separate virtual PBX's (with no default context) and two trunks coming
into separate contexts, i.e. one for residence and one for my at-home
business.
I
This might be a frequently asked question, but how do new sounds get
added to the release?
I was trying to do a popcorn extension on my phone (that gives the
date and time... maybe even getting fancy and adjusting for the
caller's timezone based on country code or area code)... but
didn't have
If the hunt-group is properly done, you should be able to busy-out
members of a trunk for maintenance.
Otherwise, if the individual trunks have numbers (unpublished) assigned
to all the circuits in the group, you could always send a Redirect() to
that any of the other trunks' numbers.
-Philip
Anyone have an application to robo-dial an outgoing conference call? ;-)
You could tie up all your circuits with outbound calls...
If you hairpin them at the switch, you shouldn't incur any usage costs...
Steve Totaro wrote:
To answer the question, there is currently no way to busy out a
guides at http://spc.pifiu.com
On Nov 18, 2007 10:53 PM, Philip Prindeville
[EMAIL PROTECTED] wrote:
I'm using a bunch of SPA942's, and I'm trying to provision them mostly
by DHCP (and what I can't set that way, I try to provision via HTTP
interface into the phone).
I changed the domain
Johansson Olle E wrote:
19 nov 2007 kl. 04.53 skrev Philip Prindeville:
I'm using a bunch of SPA942's, and I'm trying to provision them mostly
by DHCP (and what I can't set that way, I try to provision via HTTP
interface into the phone).
I changed the domain in my AstLinux config from
Yeah, I posted several hours ago and I haven't seen mine either.
-Philip
Jesse Molina wrote:
I tried re-sending my previous messages, but they are not coming
through. There is definitely some kind of filtering going on with this
list.
I like the Report website-related issues to the
I'm using a bunch of SPA942's, and I'm trying to provision them mostly
by DHCP (and what I can't set that way, I try to provision via HTTP
interface into the phone).
I changed the domain in my AstLinux config from astlinux to
redfish-solutions.com, and set
that in my sip.conf file as well:
Alan Lord wrote:
Steve Edwards wrote:
snip /
Examples of what I'd like to see:
1) A SIP telephone registering successfully.
2) A SIP telephone failing to register for reasons x, y, and z.
snip /
I'm sorry but I don't see this as being very hard. Just install
Wireshark and do
For someone that's network-aware, but hasn't sat down and plowed through
umpteen SIP-related RFC's and memorized the standards, is there a good
primer on troubleshooting SIP issues?
I'm seeing a lot of NOTIFY/603 messages on my network between Asterisk
and my Sipura 942's, for instance...
Not
That's really a question for [EMAIL PROTECTED]
The short and generally not very helpful answer is that there are a lot
of poorly packaged software releases out there that don't play well with
cross-development environments.
-Philip
Douglas Garstang wrote:
I'm trying to build an Asterisk rpm
Has anyone had any experience in getting Asterisk to interoperate with a
Coppercom switch using SIP, either as subscriber lines or else as a
trunked configuration?
And if so, do you have any configs you could share (for both ends)?
Thanks,
-Philip
Erik Anderson wrote:
On 10/20/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
If you are trying to use non-complied (XML) profiles... don't even
bother wasting your time.
Why is that? I'm using the xml-style config and they're working just fine.
I'd like to be able to
That a refrigerator is getting power is not the same as it operating
nominally.
Doors get left open... compressors fail... refrigerant eventually leaks
out of seals and coils...
Best to query it for temperature... and at a point faraway from the
coils, such as the top of the door... which
Jay R. Ashworth wrote:
On Fri, Oct 12, 2007 at 11:09:59PM +0200, F6HQZ wrote:
Check if you have a ground loop.
If yes, this is probably the cause of this hum.
Open the loop.
Actually, hum involving analog POTS lines is usually the result of the
line becoming unbalanced to ground.
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