Hello I've been trying to add a string to CIDNAME for incoming calls from
PSTN to tag calls so I know how to answer more appropriately. I have tried
numerous combinations to no avail and hope someone can point me in the right
direction. My context from extensions.conf is listed below.
When anyone leaves a voicemail message and email notifications are enabled
it causes the cpu to go to consume 100% cpu indefinetly. Note that when
email notifications are not enabled, the issue is resolved. I have been able
to re-create the circumstances on every Asterisk
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
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
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
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
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
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
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
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
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
Druid - Open Source Unified Communications
DID: +1-866-915-2407 ext 301
SIP/email: [EMAIL PROTECTED]
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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
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.
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
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
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
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
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
Adam Moffett wrote:
In all seriousness, my requirements were a little silly. A Cisco router
can fail just as a netgear router can. But I think we would find Cisco
failures to be statistically less likely.
I also think we can agree that not all devices of a certain type are
created
Gordon Henderson wrote:
On Thu, 14 Feb 2008, Phil Knighton wrote:
[softoption-zap]
exten = _0[123456789].,1,NoOp(${EXTEN})
exten = _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
exten = _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
exten =
From the RFC:
Even though the actual sampling rate for G.722 audio is 16,000 Hz, the
RTP clock rate for the G722 payload format is 8,000 Hz because that
value was erroneously assigned in RFC 1890
http://www.faqs.org/rfcs/rfc1890.html and must remain unchanged for
backward compatibility. The
as they are (and expensive -- the WIP-330 retails for $229 at
voiplink.com), I was hoping this would be a thread about simply cordless
IP (SIP or IAX) phones. I think these tend to be available at a more
reasonable price.
I have a Panasonic GLOBALRANGE BB-GT1500CB
(http://www.panasonic.ca/english
?
Unfortunately (as far as I'm aware) this is a bug in the 1.4.17 release, and
therefore I don't think any config changes will fix it. We've been told to
roll back to our previous 1.4.13 installation. It also seems to manifest
itself in ghost ringing as I've called it; place a call to a SIP
SIP wrote:
We've just launched the beta of a free service which is, really, still
only JUST out of the alpha stages.
http://www.voipmagnet.com
The basic idea is this: it's an opt-in directory focused on VoIP contact
info (with elements of social networking and privacy control).
Again
Ian wrote:
Just testing to see if my emails to this mailing list gets through.
Tried posting a question, but it failed
Thanks
Ian
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To
bilal ghayyad wrote:
Hi List;
Anyone tried to use speex, ilbc and g729 and come back
with a preferred one in the quality?
Regards
Bilal
Never miss a thing. Make Yahoo your home page.
We've just launched the beta of a free service which is, really, still
only JUST out of the alpha stages.
http://www.voipmagnet.com
The basic idea is this: it's an opt-in directory focused on VoIP contact
info (with elements of social networking and privacy control).
Again, the service is very
of course, that assumes you're reading the variable in the AGI.
SIP wrote:
Use the Set(TZone=blah) command in the dialplan. I.e.
Set(TZone=EST5EDT)
N.
Nitesh Divecha wrote:
Hello All,
Is there any way to change the timezone on the fly? I have this
little time clock program
Use the Set(TZone=blah) command in the dialplan. I.e. Set(TZone=EST5EDT)
N.
Nitesh Divecha wrote:
Hello All,
Is there any way to change the timezone on the fly? I have this little
time clock program running on Asterisk system developed using PHPAGI.
Currently, whenever user logs in,
Steve Totaro wrote:
On Jan 18, 2008 11:00 AM, SIP [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
We've just launched the beta of a free service which is, really, still
only JUST out of the alpha stages.
http://www.voipmagnet.com
The basic idea is this: it's an opt
Tzafrir Cohen wrote:
On Wed, Dec 19, 2007 at 02:47:39PM -0800, Steve Edwards wrote:
This only works because you are closed to the alternative. The alternative
(verb-noun) works fine for the above referenced applications and many more.
Do you want to tally the number of users of
Henrik Buchholz wrote:
Am Mittwoch, den 05.12.2007, 17:14 -0600 schrieb JR Richardson:
I use SIP-Realtime to store my SIP-users and I keep the informations
about the SIP-Providers my Asterisk registers to in sip.conf.
I'm running into the following problem. If I set rtcachefriends=yes
Steve Totaro wrote:
SIP wrote:
Steve Totaro wrote:
Alex Balashov wrote:
I'm sure this has been asked a million times before, but is there an easy
wa to have Asterisk register more than one (distinct) contact binding
concurrently?
The goal is to have two
-- not
the least of which being able to freely log in from anywhere at anytime
with multiple phones (the wifi sip phone from the coffee shop, the desk
phone at the office, the phone at home, the new phone I just picked up
at lunchtime) without having to configure a device entry for each and
every
at the iaxclient homepage,
There are iaxcomm, loudhush, kiax, mediax , diax and many more,
(you could also easily make your own).
Cheers,
Zoa
Vincent wrote:
On Fri, 30 Nov 2007 09:52:59 +0100, randulo [EMAIL PROTECTED]
wrote:
I have used SIP and IAX for about three years now
randulo wrote:
OK, I installed LimeSurvey and made up a new form.
http://winemailserver.com/survey/limesurvey/index.php?sid=94673
The account at the esurveryspro was deleted (not by me!) so there are
no results for that.
If anyone still has the patience to do this again, please go ahead.
We've used the Grandstream video phone quite a bit, and I have to say,
I'm considerably impressed with its quality.
YES, it's a Grandstream (and has the usual quirks and annoyances that
one has come to expect now and again), but the quality of the screen and
camera are both excellent, and with
Older base packages (older MySQL, etc).
As far as overall running Asterisk, you're not liable to run into
anything negative on the 4.5 side as opposed to 5.
N.
Zaheer K. Master wrote:
OK Thanks!
If I'm building a new Asterisk system from scratch, is there any downside to
using CentOS 4
For general SIP understanding, there's also Sip Scenario from IPtel (
http://www.iptel.org/~sipsc/ ). It will generate sort of human-readable
web stuff from captures, allowing you to click on the graphical portions
of the call and see the actual SIP packets that correspond to that.
N
Alejandro Cabrera Obed wrote:
Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server
connected to Twinkle and X-Lite clients. I have to use the GSM codec for
all of my clients, and it was set up in the sip.conf specifically in
allow=gsm line.
Twinkle has GSM codec built in,
straightforward. SER
accepts the registration and the calls, and when it needs to forward
something to Asterisk, you just add a forwarding/rewrite block to point
to the extension(s) on Asterisk you need the SIP messages to go.
Some caveats (which may be different for OpenSER, so someone else can
chime
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.
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
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
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
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,
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
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
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
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
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
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
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
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
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
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
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
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
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,
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
Is it possible to record or playback a video file in Asterisk?
N.
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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
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
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.
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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
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
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
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 .
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
and as everyone knows some
countries here has taboo to VOIP. Im not able to get phy phones registered
to my PBX as they are blocking SIP and IAX2. Hence im looking for a VPN
solution.
For this first i need to setup VPN on my server .. Am i right? Well if
anyone has experience in the whole setup how
That just seems really, REALLY dumb for a program of this magnitude.
I know this has been patched here and there by one person or another,
but does anyone know if any of these patches to make CODEC negotiation
actually, you know, negotiate a CODEC will ever make it into the core src?
Anthony Francis wrote:
Eugen Rogoza wrote:
Hello,
I'm trying to match a number in international format, like +49...
The regexp string ^\+49 doesn't work. Both in $[+49... : ^\+49]
and ${REGEX(^\+49 ${NUMBER})}.
The error is: WARNING[12486]: func_strings.c:138 regex: Malformed input
I've gotten SIP calls to work via hotspots on a Dell Axim running
SJ-Phone. I've also had reasonable success with a Nokia E60. I've had
ZERO luck from a hotspot on the UTStarCom phones, nor on the Linksys
wifi phones. I'm not quite sure yet why something which is ONLY a wifi
phone has more
the office, it really doesn't hurt to try and consolidate your
Mobile and SIP service in one of the Nokias that support it (granted, if
you're in the US, you'll have to buy them elsewhere, as the US Mobile
providers have done their best to avoid using any sort of WiFi-capable
GSM phones
Frederico,
Gizmo Project is a US company.
I hate to tell you, but under US law, they HAVE to be able to record not
only detailed CDRs about your call, but also your DTMF codes and full
conversations. Now, this would only be in the case of a warrant issued
by the US Government to retrieve
[EMAIL PROTECTED] wrote:
Hi,
I am looking for hardware sip phone with very good sound quality. Can anyone
recommend ?
I use to have Grandstream Budge-Tone 100 but I feel that the sound is not
very
satisfactory and volume too soft
Regards
ASLAY
Joe acquisto wrote:
Having had various issues with local vendor (begins with V). am looking to
move to all wireless. Anyone know if current vendor can refuse to port the current land
line numbers to a wireless provider?
From what I've read, the Fed's seem to say no, they cannot refuse, or
Joshua Colp wrote:
Alex Lake wrote:
I understand that it is customary for SIP User Agents to send OPTIONS
packets every now and then to check that a peer is still alive and
well. Indeed I understand that Asterisk itself sends them if qualify
is set to yes in the peer configuration.
How
,GotoIfTime(23:46-23:59|*|*|*?toobad)
exten = 1,n,Dial(SIP/techsupport)
exten = 1,n,GotoIf($[${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
] 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
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