No one else would step up to the plate so I
did.
Here is your list.
HAVE SOME FUN!
need any more just ask.
Troy Settle wrote: With all the discussion about licensing
issues and the sort, I think it's time for a full blown 3rd party
application to work with Asterisk while at the same time not
the link is at www.pawbell.com
- Original Message -
From: sip
To: [EMAIL PROTECTED]
Sent: Thursday, October 02, 2003 8:57 AM
Subject: Call it Asterisk-Addons and let us go have some
fun?
No one else would step up to the plate so I
did.
Here is your list.
HAVE SOME FUN!
need any
Look at www.pawbell.com they have the frontend. They
even have the NAT problem fixed!
- Original Message -
From:
23
To: [EMAIL PROTECTED]
Sent: Thursday, October 02, 2003 12:01
PM
Subject: [Asterisk-Users] Front end
Hi,
Can anyone help mewith a few links to
5volunteers needed to test NAT Transversal
software in realtime enviroment. Must be behind a firewall.
Reply to [EMAIL PROTECTED] if you would like to join
the test.
This message was checked by MailScan for WorkgroupMail.
www.workgroupmail.com
Everyone seems to be working on their own servers
that are in there homes, offices and elsewhere. The only common thread is this
list-serv.
I haveseveral * servers set up here in
Austin, Texas. I propose we set up one of them with with all the list-serv
members. At this time the calls would be
IT!!!
- Original Message -
From: WipeOut [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, October 04, 2003 8:10 AM
Subject: Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
sip wrote:
Everyone seems to be working on their own servers that are in there
homes, offices and elsewhere
will
be having voice messages and realtime talks with other programmers and
developers.
I think this should set a precedent for other mailing list also.
Rgds
Manoj K Gupta
- Original Message -
From: sip [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, October 04, 2003 7:31 PM
IT!!!
sip wrote:
IAXTEL is a 1-700 system designed for on-net calls.
We have very low-cost PSTN lines to 48 states for no-cost long-distance
dialing. I plan to add some of these lines to the system. As servers are
added in other cities around the world I envision dialing Berlin, Germany
from Austin
But it would be a free call to the common man who had a fast internet
connection and a softphone or IP phone. He doesn't have to have a server or
know the tech stuff... just a free softphone and he is in.
After all, we are all working to develop this industry...build
servers...sell phones...etc.
count me in
- Original Message -
From: Paulo Mannheimer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, October 17, 2003 12:23 PM
Subject: [Asterisk-Users] Beta testers for visual configuration tool for
asterisk
Hi All,
We've been developing for a while an IDE for Asterisk, and
to apologize.
-- Forwarded message --
From: voiplist [EMAIL PROTECTED]
Date: Jul 31, 2007 10:50 PM
Subject: Live Answering Service with Direct SIP Connections and Light
Accounts starting at $14.95/mo
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
STUN is a pretty simplistic server. There's nothing else that needs to
be configured on the STUN server side. It's pretty much either running
or it's not.
Just start plugging in the server to your clients and give it a whirl.
It should work.
N.
Rizwan Hisham wrote:
Hi all,
This is the
?
On 8/1/07, *SIP* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
STUN is a pretty simplistic server. There's nothing else that needs to
be configured on the STUN server side. It's pretty much either running
or it's not.
Just start plugging in the server to your clients
IdeaSIP, Voxbone, Gizmo Project, etc... if we're talking retail.
N.
Mail list wrote:
I am looking for a retail DID provider which should provide unlimited
incoming calls something around 4-5 bucks . Nufone seemed like a good
choice at $5 but they are down again :( . Any suggestions please .
will then know the proper IP address to use to send data back to the UA.
This is primarily of importance when you are using SER/OpenSER as a SIP
proxy, or have Asterisk set to canreinvite=yes
What happens is that this allows clients to directly talk to each other
using publicly-addressable IP
Stephen Bosch wrote:
Douglas Garstang wrote:
I confused by this. Don't ITSP's have redundancy? Don't they have
multiple edge systems for accepting incoming calls? Don't their multiple
edge systems have multiple interfaces, connected to multiple subnets,
via multiple switches? And, don't
Trevor G. Hammonds wrote:
From: SIP
Sent: Saturday, August 04, 2007 2:57 PM
Stephen Bosch wrote:
Douglas Garstang wrote:
I confused by this. Don't ITSP's have redundancy? Don't they have
multiple edge systems for accepting incoming calls? Don't their multiple
edge
John Novack wrote:
Paul wrote:
The thread is about music on hold. Things such as playing local radio
stations in a waiting room are not related. I don't think there is anything
illegal about using normal over the air radio and TV for such purposes as
long as it stays in the local
Worthless comes in many forms, Doug. If you're talking specifically
about the monetisation of hardware/effort, then it may indeed be
worthless by the simple fact that the cost may outweigh the net gains in
profits gained from the purchasing, configuration, and deployment.
Businesses are about
Jay R. Ashworth wrote:
On Sun, Aug 05, 2007 at 07:28:05PM -0400, SIP wrote:
Lots of information around about people who've had issues with
rebroadcasting the radio in their business establishments. However, it
is rare that ASCAP et al go after anyone but the big moneymakers. The
old
Jay R. Ashworth wrote:
However, if you get caught willfully performing copyrighted music
without paying ASCAP, BMI, et al, you're liable for a $100,000 fine
($20,000 per song if it's not deemed willful) per song.
I wonder how much of *that* money goes to the songwriters. ;-)
Cheers
, but
about the hideous delay caused by the sat latency. ;)
All SIP deployments should come with emergency communications kits
consisting of two cans and a spool of string.
N.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com
I believe X-Lite v3 (and EyeBeam) from Counterpath both support 3856 as
presence user agents.
N.
Kate Kretz wrote:
sorry, I meant RFC 3856, sip presence, not sip regitration
On 8/7/07, *Tzafrir Cohen* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
On Tue, Aug 07, 2007 at 08:16
Steve Totaro wrote:
Matt wrote:
Did I miss something? I see Digium no longer contracts with Allison
to record IVR prompts, was there a falling out?
Where do you see that?
Thanks,
Steve
http://www.digium.com/en/products/voice/
She's still on the website.
N.
, that indicates less a cessation of contract with Digium/Allison
and more a modification of the way things are handled. But who knows.
N.
Matt wrote:
She's sort of on the website... click 'Purchase and Price', then 'Buy
Online', You will see there is no place to purchase it.
On 8/9/07, SIP
Is it possible to record or playback a video file in Asterisk?
N.
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To UNSUBSCRIBE or update options visit:
Jay R. Ashworth wrote:
On Thu, Aug 16, 2007 at 11:00:05PM -0400, Zeeshan Zakaria wrote:
This is really ridiculous. So this means that now nobody can use
fax-to-email without paying to J2 first?
Horseshit. Prior art is trivial. How old is Hylafax?
Cheers,
-- jra
It's
Russell Bryant wrote:
Steve Murphy wrote:
How about this one: from an extensions.conf that someone posted on the
internet, I think, and I converted to AEL; I'm sorry, but I can't find
the original author.
(If anybody can find his post, I'd love to give him credit.) I did test
this out,
Asterisk is an excellent PBX system, and makes a very good endpoint in
the SIP chain for all sorts of things -- IVR systems, voicemail
applications, automated messages, etc.
It has an extremely well-written CDR engine, so many people mesh it with
billing applications to produce accurate
Lundmark wrote:
I'm still learning myself, but SEMS (iptel.org/sems) seems to offer
many of the media- and/or b2bua-functions that Asterisk do.
///Fredrik
- Original Message - From: SIP [EMAIL PROTECTED]
To: Nhadie [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com; [EMAIL PROTECTED
Tzafrir Cohen wrote:
On Fri, Aug 24, 2007 at 10:05:44PM -0500, Carlos Leal wrote:
Launched the OS X version of Gizmo after about a year of inactivity,
downloaded the update and discovered the new improved Giszmo features
Asterisk interoperability by allowing a secondary SIP account
Jay R. Ashworth wrote:
2) Asterisk will attempt to complete a call (rather than correctly
returning reorder) when it can't allocate a codec for both directions
of the call.
Yes, Asterisk will complete the call and you will have no audio if you
have no free licenses.
Ok; am I
This is actually a big misconception... the idea that you don't need to
match + because you'll never receive a + and it's just a metacharacter.
In the modern world of IP phones and such, more often than not, you will
ACTUALLY be sent a + and will need to translate that yourself on your
own
Snom, UTStarCom, and the usual assortment of softphones (X-Lite,
SJPhone, Snom360 Softphone, eyeBeam, Bria).
N.
Anthony Francis wrote:
What phones are you using?
SIP wrote:
This is actually a big misconception... the idea that you don't need to
match + because you'll never receive
the web interface
(many of our users do that, and they just type a + like a normal human)
or you can dial by the phone keypad in which case + is available in the
same meny with the @ symbol (remember, these are SIP-capable phones --
without an @ symbol, it's a poorly designed phone).
From
Anthony Francis wrote:
SIP wrote:
(many of our users do that, and they just type a + like a normal human)
I don't know if you intended to be rude with the normal human comment but
it sure seems like it when reading your reply. Also how many users know they
can dial ** to get
Actually, when someone asked a legitimate question about how to account
for a + sign, you jumped into the thread saying that such things were
useless and that people should just learn how to dial 011 since that's
all a plus sign means. When it was pointed out that it IS, in fact, a
legitimate
Adrian Marsh wrote:
All,
I've been experimenting with shortcodes for SIPgate etc. Passing calls
to SIPbroker seems a good way to go, but the message I've had back from
SIPgate is we don't support SIPBroker...
So whats the easiest way to support SIP SIP network calling?
At the moment
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of SIP
Sent: 04 September 2007 15:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIPBroker vs SIPgate
Adrian Marsh wrote:
All,
I've been experimenting with shortcodes for SIPgate
Curses! I just got BACK from Vienna yesterday. I should have stayed
another week. :)
N.
Klaus Darilion wrote:
Hi!
I proudly announce the first ser/openser/asterisk beer drinking evening
in Vienna.
When: Thursday (thirsty day) 20. September 2007, 19:00 CEST
Where: Vienna, a bar in an
Not at all relevant to your query, but I still use the mysql CLI for any
mysql task... and while most OSs have nice, functional tools to add
users (command-line tools), there are SOME (*cough* Irix *cough*) where
there are no CLI tools and VI is your only option (especially if you're
remotely
Jim Canfield wrote:
SIP wrote:
Not at all relevant to your query, but I still use the mysql CLI for any
mysql task... and while most OSs have nice, functional tools to add
users (command-line tools), there are SOME (*cough* Irix *cough*) where
there are no CLI tools and VI is your only
Now see... the fact that it can't handle 6 DS3s doesn't mean it's not a
softswitch... just that it's not a carrier-grade softswitch.
N.
Alex Balashov wrote:
Perhaps I'll be a little more amicable when someone finds a way to bring
at least five or six DS3s into Asterisk.
On Wed, 19 Sep
Per Jessen wrote:
Atis Lezdins wrote:
This seems nice way of sharing settings, however it wouldn't take over
calls in progress. For us, currently the greatest problem is that
whenever Asterisk crashes, calls are lost, and that means - lost
money. Are there any ideas?
Perhaps
Probably.
Andreas van dem Helge wrote:
So I'm the only person that actually enjoys reading the RFC's?
On 10/7/07, Brian West [EMAIL PROTECTED] wrote:
Telling someone to read the RFC bah.. might as well give them a blanket and
pillow because they will fall asleep. chan_sip is just ugly
Russell Bryant wrote:
I have been having discussions with various members of the development
community
in regards to changes to the way we manage open source Asterisk releases. The
changes that we eventually decide on will determine how we manage the 1.6
version of Asterisk. I will be
Nonsense! I'm a Security Expert (TM) and I say run EVERYthing on your
firewall
And...uh... what was your IP again? ;)
N.
Steve Prior wrote:
GNUbie wrote:
By the way, my Asterisk PBX server is also my wireless access point,
web server, file server, music server, VPN server,
Drew Gibson wrote:
Tzafrir Cohen wrote:
On Thu, Oct 18, 2007 at 01:04:06PM +0100, Cartwright, Dave wrote:
Just for fun.
http://news.bbc.co.uk/1/hi/magazine/7049642.stm
It's Asterix != Asterisk. Though named after *.
In Britain, it's called humour :-)
regards,
Drew
Direct single line video conferencing via SIP is actually pretty
straightforward and works rather well.
Multipoint conferencing is where you get into a bit of a mess. There
are precious few products out there that claim multipoint SIP video
conferencing capability, and we've had no luck so
Dominic Son wrote:
Could you please call it and confirm with me it's not working for you
either? I should probably transfer my DID number anyways, if I could
only get them to respond! Does anyone have a suggestion as to where to
go in this situation? Possibly a place with high capacity
Tilghman Lesher wrote:
On Friday 26 October 2007 16:13:11 Douglas Garstang wrote:
Thanks. I am quite familiar with ngrep. I was asking how I could get the
SIP response code from the dial plan.
You cannot. And you shouldn't have to. The dialplan should be generic
to all protocols
Gordon Henderson wrote:
On Mon, 29 Oct 2007, Abdul wrote:
Hi,
Is it possible to have multi listening bindport in asterisk?
Now days mostly ISPs are Blocking the standard 5060 port so we want to
keep option if 5060 is blocked we can ask our customers to use another
port.
I'm pretty sure he's asking what sort of advantages there are in using
VoIP (and probably Asterisk) over traditional wireline services.
Advantages being things like flexibility and portability (with cost and
barriers-to-entry being somewhat debatable). But he's more interested
perhaps in the
care.
In the grand scheme of things, phone are cheap. With SIP phones, employees
can move their phone to another office if they move and just plug it in.
Companies can also better monitor employees.
My mobile phone supports SIP (via WiFi) 3G and GSM... So I can move about
Lacy Moore wrote:
On Tue, Mar 25, 2008 at 11:19 AM, Jiffy Slides Leonard Burton
[EMAIL PROTECTED] wrote:
HI,
We need to get our number into the White Pages.
Has anyone here actually tried it?
It's not just Voip numbers. We've got a PRI from XO that (even though
they say
Mike Trest - On Travel wrote:
At 01:17 PM 4/22/2008, you wrote:
My question would be - is this actually compliant with the FCC E911
regulations applicable to VoIP providers?
IMHO and EXPERIENCE before FCC, this arrangement is NOT compliant
Reason: multiple subscribers using the
Jaap Winius wrote:
Quoting Jerry Harshany [EMAIL PROTECTED]:
There is an additional alternative for a ringback to a caller, which
is to use the Call File capability as noted in Van Meggelen's
Future of Telephone; 2nd ed, p306.
As it says in the book, call files allow calls
Tilghman Lesher wrote:
We're about to do another batch of sounds, and I see by my word count that we
have some extra time left over. So, suggestions will be entertained for
additional prompts in English, Spanish, or French. The only rules are: 1) the
prompts have to be generic to Asterisk.
Tilghman Lesher wrote:
On Tuesday 06 May 2008 06:58:39 Steve Totaro wrote:
All I see in the ABE release
notes is 1.2 although I have heard that ABE should be running 1.4
Very Soon many many moons ago
http://www.digium.com/en/docs/ABE/README . So either Digium doesn't
trust 1.4 enough
Druid - Open Source Unified Communications
DID: +1-866-915-2407 ext 301
SIP/email: [EMAIL PROTECTED]
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Joseph wrote:
I need another Sipura 3K and the replacement I think is Linksys SPA3102.
Any input on how reliable is it?
We have a few dozen subscribers using them at any given point in time. I
and my wife even use them at our respective homes. Rock solid stable.
No issues whatsoever.
N.
Dave Cotton wrote:
SIP wrote:
Joseph wrote:
I need another Sipura 3K and the replacement I think is Linksys SPA3102.
Any input on how reliable is it?
We have a few dozen subscribers using them at any given point in time. I
and my wife even use them at our respective
Is there a way to set a call timer on calls created with call files? I'm
looking specifically at having Asterisk hang up the call after a certain
period of connection.
Obviously, when I try passing an |S(time) on the channel line, I get an
invalid call file... so I'm wondering if there's
That worked beautifully. Thanks, Mark.
N.
Mark Michelson wrote:
Mark Michelson wrote:
SIP wrote:
Is there a way to set a call timer on calls created with call files? I'm
looking specifically at having Asterisk hang up the call after a certain
period of connection.
Obviously
When calling from our SIP proxy through Asterisk to the PSTN provider,
we support reINVITES which tend to work seamlessly.
However, when creating a call file which essentially connects a call
from the SIP proxy to the SIP proxy, Asterisk wants to stay in the RTP
media path. I understand
Gonzalo Servat wrote:
On Thu, Aug 7, 2008 at 2:04 PM, Joseph [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I just received an email notice from FWD about $30 membership fee.
My question: Is the email genuine? Did anybody else receive it?
I'm just trying to be sure that it
SIP wrote:
When calling from our SIP proxy through Asterisk to the PSTN provider,
we support reINVITES which tend to work seamlessly.
However, when creating a call file which essentially connects a call
from the SIP proxy to the SIP proxy, Asterisk wants to stay in the RTP
media path. I
Russell Bryant wrote:
On Aug 11, 2008, at 12:04 PM, SIP wrote:
SIP wrote:
When calling from our SIP proxy through Asterisk to the PSTN
provider,
we support reINVITES which tend to work seamlessly.
However, when creating a call file which essentially connects a call
from
test? Presumably, if I had
this, I could rent a PSTN number from a US-based provider, and point it to
the appropriate SIP/IAX address. I expect my total usage would be just a
few minutes, though having the facility available for a few weeks would be
helpful, to allow me to play around
Tzafrir Cohen wrote:
On Wed, Aug 20, 2008 at 10:00:55AM -0700, Eric Chamberlain wrote:
We are exploring using Asterisk for a project and we are looking for a
way to encrypt/decrypt the peer passwords stored in the realtime
database (postrges).
Ideally, we want to use a public key to
Igor Hernandez wrote:
I was thinking the same thing I believe Tzafrir just alluded to. If the
passwords are encrypted in the DB with a public key then...asterisk
needs to have the private key stored somewhere to be able to decrypt the
values to authenticate the user. In this way there is
Jay R. Ashworth wrote:
On Tue, Aug 26, 2008 at 05:10:35PM -0400, Asterisk wrote:
The shared desktop is available using a Java enabled browser at
???http://callin.xelatec.com/vnc??? with a password of ???aretta???.
Of course you must first have Zoiper installed and then add a new Zoiper
Really? I thought both IAX and SIP are, at 3 characters apiece, equally
short.
However, if you get into IAX2, then yes... SIP is definitely a shorter
answer.
N.
Alex Balashov wrote:
The short answer is SIP.
Stefan Gofferje wrote:
http://www.voip-info.org/wiki-IAX
http://www.voip
Olivier wrote:
Hi,
A somehow related question, is broadcasting streaming music as music
on hold, submitted to any licencing fee ?
Regards
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It's common sense. Using all iLBC, I can't seem to get 100 simutaneous
calls on my AMD 486 dx2/66.
I don't get it! ;)
N.
Eric ManxPower Wieling wrote:
Where did you hear this?
Shaun Wingrin wrote:
I have heard it said that, Asterisk falls over at 100 simultaneous
calls. Is this
My thoughts are that to do parallel requests from every IP address on
the machine is extremely weird behaviour.
How would any server know which to respond to?
SIP forking is supposed to send requests to multiple different
destinations (or fork mid-stream to send to different destinations
of packets. Figure out what to do with them. I'll be waiting for
your response.
There's a reason routing rules exist and mature services allow you to
control the interface from which it originates.
N.
Brian J. Murrell wrote:
On Thu, 2008-09-25 at 14:56 -0400, SIP wrote:
Sending from
Alex Balashov wrote:
You need to define what you mean by SIP forking. There are many
things people mean by that. They are usually one of:
1) Call branching (proxies do this).
2) Parallel but distinct call legs managed by a UAC (this is what
Asterisk does when you Dial(SIP/exten1SIP
gateway, your machine is misconfigured and internet traffic will not
properly flow.
I know you're just the messenger here, and it's not your fault. But the
message is wrong. Ekiga has tried to solve a problem (that of
determining a 'best path' for SIP to allow data flow in a NAT or
filtered scenario
Brian J. Murrell wrote:
On Fri, 2008-09-26 at 10:16 -0400, SIP wrote:
The RFCs are there for a reason. All SIP forking is UAS territory. Not
UAC territory.
From http://bugzilla.gnome.org/show_bug.cgi?id=553810 Damien Sandras
asks:
I repeat, Ekiga is doing something
Eric ManxPower Wieling wrote:
Olivier wrote:
I don't have any spare zaptel enabled system I could try this on, but I
was not aware of this CHANNEL variable.
Now, I can see it here http://www.voip-info.org/wiki/view/Asterisk+variables
Maybe, I will add a line in www.voip-info.org
Philipp Kempgen wrote:
Andrew Kohlsmith (lists) schrieb:
On October 5, 2008 12:22:37 pm Philipp Kempgen wrote:
---cut---
http://lists.digium.com/pipermail/asterisk-users/2008-October/219538.html
http://lists.digium.com/pipermail/asterisk-users/2008-October/219541.html
it, but not being the world's most
proficient C coder, I'm always afraid I'll break something else. ;)
N.
Andrew Joakimsen wrote:
I am using a Polycom 501 SIP phone behind NAT. Asterisk server is
public with no NAT... everything works on the Asterisk end just fine
EXCEPT that I can never check voice mail
/OpenSIPS more
than SER.
Are there some hard reasion for this.
I am in the process of deciding which SIP server i want to use with
Asterisk and just made a go at SER. Compilation was a little rough but
it was manageable. I threw away every module which funtionality i didn't
wanted
Alex Balashov wrote:
SIP wrote:
Seriously, though... this seems to be a popular misconception. I hear it
a lot. Where did you come across the information that SER is no longer
developed?
That seems to be a consequence of looking at the releases.
Anyway, I spoke too soon
It's not 100% secure. Like any dual-key encryption, it's subject to the
classic man-in-the-middle attack. This is why the Windows Zfone app has
the addition of a visual key you can read and coordinate with the
recipient to determine if a MITM attack is occurring. But only if you
know what you're
Joseph wrote:
I'm using Linksys SPA3102 adapter and have a strange ring tone:
Long-Short-Short or Long-Long-Short-Short
Does anybody know which setting adjust this ring tone on SPA3102
Sipura rings normally. I'm not sure if setting are on Regional Tab or User Tab
Interestingly, I get
randulo wrote:
On Tue, Nov 4, 2008 at 5:00 PM, Michael Graves [EMAIL PROTECTED] wrote:
In any case, the wideband bridge for this weeks VUC call supports only
G.722.
But we do plan to make a recording of both conference version available,
AFAIK?
r
But will it be a high-def
JR Richardson wrote:
Sorry if this hit the list twice, sent out yesterday, but didn't see
it show up.
Hi All,
Can Asterisk be used as a SIP proxy, blah, blah, blah???
I've glanced over questions like this through the years, with a good
idea on
what a SIP proxy is and what Asterisk
shadowym wrote:
I have some general questions about marketing. Lot's of technical info but
I was wondering how people are getting the business to begin with. I'm from
the IT end of things but Telco is quite a bit different. Is cold calling
still the way to go or networking? General stuff
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There was a mailing yesterday about issues ASA-2007-010 (buffer overflow
in SIP T.38 code) and ASA-2007
Businesses RARELY are in a position to choose new Telco systems
providers. Oftentimes, that sort of decision is made by whomever leases
them the office space, or was made once back in the beginning, and
they've had no real reason to re-evaluate their service/provider. There
are, however,
$25/minute? Souds pretty cheap. We've seen PRS numbers in the area of
$500/minute for PRS fraud.
N.
Steve Totaro wrote:
I suspect that this will happen more and more. I also suspect that many
people who have weak SIP credentials like user=100 secret=100 will be
the victim of toll fraud
] On Behalf Of SIP
Sent: Wednesday, April 25, 2007 6:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FYI
$25/minute? Souds pretty cheap. We've seen PRS numbers in the area of
$500/minute for PRS fraud.
N.
Steve Totaro wrote:
I suspect
PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of SIP
Sent: Thursday, April 26, 2007 8:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FYI
Somalia. Nigeria. We've seen it as high as $250/minute in Egypt.
Probably others.
Steve
Premium Rate Services think like 900 and 976 numbers in the US, but
not every country allocates a particular block of numbers or prefixes to
its premium rate services, so with some, they're pretty close to
impossible to block.
Stephen Bosch wrote:
SIP wrote:
It can and it has
,GotoIfTime(23:46-23:59|*|*|*?toobad)
exten = 1,n,Dial(SIP/techsupport)
exten = 1,n,GotoIf($[${DIALSTATUS} = BUSY]?toobad)
exten = 1,n,Hangup
exten = 1,n(toobad),VoiceMail([EMAIL PROTECTED])
Very messy. Alternatively:
exten = 1,1,GotoIfTime(0:00-0:00|*|*|*?woohoo)
exten = 1,n,GotoIfTime(0:15-0:15
Joshua Colp wrote:
Alex Lake wrote:
I understand that it is customary for SIP User Agents to send OPTIONS
packets every now and then to check that a peer is still alive and
well. Indeed I understand that Asterisk itself sends them if qualify
is set to yes in the peer configuration.
How
Joe acquisto wrote:
Having had various issues with local vendor (begins with V). am looking to
move to all wireless. Anyone know if current vendor can refuse to port the current land
line numbers to a wireless provider?
From what I've read, the Fed's seem to say no, they cannot refuse, or
[EMAIL PROTECTED] wrote:
Hi,
I am looking for hardware sip phone with very good sound quality. Can anyone
recommend ?
I use to have Grandstream Budge-Tone 100 but I feel that the sound is not
very
satisfactory and volume too soft
Regards
ASLAY
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