On Tue, Feb 28, 2012 at 8:28 PM, Alejandro Imass a...@p2ee.org wrote:
What you are saying seems impossible and makes no sense unless the
router is assigning a public IP or is SIP aware and knows how to
read the routing data contained inside the SIP packets, and none of
the consumer routers are
On Tue, Feb 28, 2012 at 8:58 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
[...]
Yes, I have had no problems with Grandstream first gen ATAs, configured with
server and credentials and shipped off, they just work.
We use the HT-286, the server is on a public IP the nat setting on
On 02/29/2012 08:22 AM, Alejandro Imass wrote:
We use the HT-286, the server is on a public IP the nat setting on
asterisk is set to yes and without port re-direction the ATAs have
never connected from a private network, so I honestly find this SIP
plug and play very hard to believe. But if it
On Wed, Feb 29, 2012 at 9:44 AM, Kevin P. Fleming kpflem...@digium.com wrote:
On 02/29/2012 08:22 AM, Alejandro Imass wrote:
[...]
The number of 'plain' SIP endpoints deployed behind consumer-grade NAT
devices talking to Asterisk servers on public IP addresses is in the
millions, if not the
On 02/29/2012 09:25 AM, Alejandro Imass wrote:
On Wed, Feb 29, 2012 at 9:44 AM, Kevin P. Flemingkpflem...@digium.com wrote:
On 02/29/2012 08:22 AM, Alejandro Imass wrote:
[...]
The number of 'plain' SIP endpoints deployed behind consumer-grade NAT
devices talking to Asterisk servers on
On Wed, Feb 29, 2012 at 9:22 AM, Alejandro Imass a...@p2ee.org wrote:
On Tue, Feb 28, 2012 at 8:58 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
[...]
Yes, I have had no problems with Grandstream first gen ATAs, configured
with
server and credentials and shipped off, they just
On Wed, Feb 29, 2012 at 8:34 AM, Kevin P. Fleming kpflem...@digium.com wrote:
Certainly there are plenty of examples of SIP endpoints working poorly
behind NAT devices, and replacing that endpoint with an IAX2 endpoint curing
the symptoms. Invariably, this is caused by the fact that the NAT
On Wed, Feb 29, 2012 at 9:44 AM, Kevin P. Fleming kpflem...@digium.comwrote:
On 02/29/2012 08:22 AM, Alejandro Imass wrote:
We use the HT-286, the server is on a public IP the nat setting on
asterisk is set to yes and without port re-direction the ATAs have
never connected from a private
On Wed, Feb 29, 2012 at 8:41 AM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
Agreed with one exception, the endpoint behind the NAT DOES need to be setup
correctly to keep the router from seeing inbound traffic to the device as
unsolicited and drop it. That is a function of the router but
On Wed, Feb 29, 2012 at 10:43 AM, Carlos Alvarez car...@televolve.comwrote:
On Wed, Feb 29, 2012 at 8:41 AM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
Agreed with one exception, the endpoint behind the NAT DOES need to be
setup
correctly to keep the router from seeing inbound
On Wed, Feb 29, 2012 at 8:58 AM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
So you turned it off on the phones but use it on the Asterisk side?
Do you set a value or just use qualify=yes?
Yes, just as I said, just qualify=yes.
Did you submit a bug report? If it is easy to reproduce
On 2012-02-28 21:22:44 +, Kevin P. Fleming said:
On 02/28/2012 03:08 PM, Troy Telford wrote:
[myprovider]
type=friend
username=
secret=
context=somecontext
host=provider_server
qualify=1000
disallow=all
allow=g729
allow=ulaw
auth=md5,rsa
requirecalltoken=yes
trunk=yes
A serious bug with
On 02/29/2012 11:35 AM, Troy Telford wrote:
On 2012-02-28 21:22:44 +, Kevin P. Fleming said:
On 02/28/2012 03:08 PM, Troy Telford wrote:
[myprovider]
type=friend
username=
secret=
context=somecontext
host=provider_server
qualify=1000
disallow=all
allow=g729
allow=ulaw
auth=md5,rsa
On 2012-02-29 15:25:49 +, Alejandro Imass said:
We use SIP and IAX interchangeably, but had less hassle with IAX. The
topic of the discussion on this thread was that SIP is so awesome and
that IAX is a peice of crap.
The original question (mine) was that my sound quality when using IAX
On Wed, Feb 29, 2012 at 10:34 AM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
[...]
If you can post some SIP debug info from an ATA trying to register without
any redirection and also the relevant portions of your sip.conf, I am sure I
can help.
Do it from a new location with an el
On Wed, Feb 29, 2012 at 1:05 PM, Troy Telford ttelford.gro...@gmail.com wrote:
On 2012-02-29 15:25:49 +, Alejandro Imass said:
We use SIP and IAX interchangeably, but had less hassle with IAX. The
topic of the discussion on this thread was that SIP is so awesome and
that IAX is a peice of
On Wed, Feb 29, 2012 at 1:26 PM, Alejandro Imass a...@p2ee.org wrote:
On Wed, Feb 29, 2012 at 1:05 PM, Troy Telford ttelford.gro...@gmail.com
wrote:
On 2012-02-29 15:25:49 +, Alejandro Imass said:
We use SIP and IAX interchangeably, but had less hassle with IAX. The
topic of the
On my Asterisk system, I'm using a provider that provides both IAX2 and
SIP connectivity.
Personally, I'd prefer to use IAX2, and that's what my account is setup
to use. However, I'm having a problem:
With IAX2:
- Incoming Voice from my Provider - Asterisk = Sounds great
- Outgoing Voice
@lists.digium.com
Subject: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On my Asterisk system, I'm using a provider that provides both IAX2 and SIP
connectivity.
Personally, I'd prefer to use IAX2, and that's what my account is setup to use.
However, I'm having a problem
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Troy Telford
Sent: Tuesday, February 28, 2012 3:08 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On my Asterisk system, I'm using a provider that provides both IAX2
On 02/28/2012 03:08 PM, Troy Telford wrote:
[myprovider]
type=friend
username=
secret=
context=somecontext
host=provider_server
qualify=1000
disallow=all
allow=g729
allow=ulaw
auth=md5,rsa
requirecalltoken=yes
trunk=yes
A serious bug with IAX2 trunking in recent versions of Asterisk (you did
.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Troy
Telford
Sent: Tuesday, February 28, 2012 4:08 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On my
: Tuesday, February 28, 2012 3:08 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On my Asterisk system, I'm using a provider that provides both IAX2 and SIP
connectivity.
Personally, I'd prefer to use IAX2, and that's what my account
On 2012-02-28 22:17:37 +, Danny Nicholas said:
Ok Steve, obviously you've outsmarted at least this poster. On the one
hand, IAX2 has purchased things for you (won't go as far as saying it
bought your Mercedes), but on the other hand it is being dropped by
providers as we speak. So are
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Same provider - IAX sounds bad, SIP
sounds great
** **
PSS
** **
http://bit.ly/ywiwzt
On Tue, Feb 28, 2012 at 4:56 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
Google
: Tuesday, February 28, 2012 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds
great
PSS
http://bit.ly/ywiwzt
On Tue, Feb 28, 2012 at 4:56 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote
Roger That, I am an IC. I contract with the Government to little ten phone
shops. From VA/MD/DC area, I have been contracted and flown in to many
large call center locations that were CONUS and OCONUS.
My facebook is Steve Totaro in Reston VA. LinkedIN is more accurate, but
my resume speaks
- Non-Commercial Discussion
Subject: Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds
great
PSS
http://bit.ly/ywiwzt
On Tue, Feb 28, 2012 at 4:56 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
Google or click this link http://bit.ly/ywiwzteve
On 2012-02-28 21:22:44 +, Kevin P. Fleming said:
A serious bug with IAX2 trunking in recent versions of Asterisk (you did
not mention what version you are using) was just resolved last week. You
should test with 'trunk=no' to see if that is the cause of your problem;
it seems very likely.
They said the same thing in 2005, 2008, now Every release.
You never answered the question as to why you don't want to use SIP. Is
there a reason, or do you just want to torture yourself?
Thanks,
Steve T
On Tue, Feb 28, 2012 at 6:23 PM, Troy Telford ttelford.gro...@gmail.comwrote:
On
BTW, Trunking was the other selling point of IAX2 besides using 1 port
which is easily a DDOS target and also probably still
an implantation problem of using one thread and one proc for all calls.
Trunking allowed for less overhead then SIP since all the overhead for the
concurrent calls were
IAX is not supported or taken seriously outside the Asterisk ghetto,
and that's good enough reason not to use it, IMHO.
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web:
Hey Alex,
Hope you are well.
Just a piece of advice. Many or most people do not know the real
definition of ghetto and take it as a negative, poor, racial, black,
connotation.
Your vocabulary and and ability to articulate correctly can get you in
trouble sometimes.
Anyone that thinks that the
Wow Wikipedia was the only place that had the original meaning and not the
slur or slang meaning.
A *ghetto* is a section of a city predominantly occupied by a group who
live there, especially because of social, economic, or legal issues. The
term was originally used in Venice
On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
[...]
Without trunking, you only have the single port thing. It is quite easy to
Nope. The main reason _we_ use IAX is because it's easier for NAT
open the correct ports for SIP, some just have GUI with a SIP
Oops, I meant da Asterisk 'hood. Thanks for the protip.
On 02/28/2012 06:55 PM, Steve Totaro wrote:
Hey Alex,
Hope you are well.
Just a piece of advice. Many or most people do not know the real
definition of ghetto and take it as a negative, poor, racial, black,
connotation.
Your
On 2012-02-28 23:29:53 +, Steve Totaro said:
They said the same thing in 2005, 2008, now Every release.
You never answered the question as to why you don't want to use SIP. Is
there a reason, or do you just want to torture yourself?
Probably self-torture, yes. I want to at least try
Perhaps your users live in an internet ghetto where the routers are
similar to Yugos with spinners. We haven't run into any routers that
don't do NAT properly in a very very long time.
On Tue, Feb 28, 2012 at 5:07 PM, Alejandro Imass a...@p2ee.org wrote:
On Tue, Feb 28, 2012 at 6:36 PM, Steve
On Tue, Feb 28, 2012 at 7:19 PM, Carlos Alvarez car...@televolve.com wrote:
Perhaps your users live in an internet ghetto where the routers are
similar to Yugos with spinners. We haven't run into any routers that
don't do NAT properly in a very very long time.
Perhaps you should read again
On Tue, Feb 28, 2012 at 5:35 PM, Alejandro Imass a...@p2ee.org wrote:
works. This cannot be done with SIP and off the shelf cheap ATAs,
period.
We do it, so cannot seems to be a strong word. It's not perfect,
but our IAX problems outnumbered the SIP problems by at least double.
Your mileage
On Tue, Feb 28, 2012 at 7:41 PM, Carlos Alvarez car...@televolve.com wrote:
On Tue, Feb 28, 2012 at 5:35 PM, Alejandro Imass a...@p2ee.org wrote:
works. This cannot be done with SIP and off the shelf cheap ATAs,
period.
We do it, so cannot seems to be a strong word. It's not perfect,
Please
On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass a...@p2ee.org wrote:
Please expand as to how you set-up a SIP ATA behind a common home
router set-up, without port redirection and/or use of a SIP proxy
and/or STUN server? Unless the ATA has some sort of magic (e.g. VPN
support) it _cannot_ be
On Tue, Feb 28, 2012 at 7:07 PM, Alejandro Imass a...@p2ee.org wrote:
On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
[...]
Without trunking, you only have the single port thing. It is quite easy
to
Nope. The main reason _we_ use IAX is because it's
On Tue, Feb 28, 2012 at 7:41 PM, Carlos Alvarez car...@televolve.comwrote:
On Tue, Feb 28, 2012 at 5:35 PM, Alejandro Imass a...@p2ee.org wrote:
works. This cannot be done with SIP and off the shelf cheap ATAs,
period.
We do it, so cannot seems to be a strong word. It's not perfect,
but
Roger That!
On Tue, Feb 28, 2012 at 8:28 PM, Carlos Alvarez car...@televolve.comwrote:
On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass a...@p2ee.org wrote:
Please expand as to how you set-up a SIP ATA behind a common home
router set-up, without port redirection and/or use of a SIP proxy
Just to stir the pot a bit, I am a member of a worldwide private network
of Asterisk and AstLinux users. the network uses IAX exclusively, and we
have no issues relating to audio quality with a large variety of
providers, routers, host machines, and expertise in configuration of the
specific
On Tue, Feb 28, 2012 at 8:28 PM, Carlos Alvarez car...@televolve.com wrote:
On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass a...@p2ee.org wrote:
Please expand as to how you set-up a SIP ATA behind a common home
router set-up, without port redirection and/or use of a SIP proxy
and/or STUN
- IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 8:28 PM, Carlos Alvarez car...@televolve.com wrote:
On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass a...@p2ee.org wrote:
Please expand as to how you set-up a SIP ATA behind a common home
router set-up, without port redirection
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