On 28/02/2012 7:58 PM, DHAVAL INDRODIYA wrote:
Hi All,
I have one question that if my device is registered over TLS on
asterisk .
is it required that it can only use SRTP for making an outbound calls
or incoming calls too.
No.
how we can disable srtp and only enable TLS.
Hi,
I have to make an asterisk gateway in front of several other asterisk.
This gateway will essentialy be used for outbound call.
This gateway will be connected to other asterisk by IAX trunk, outbound
call will use SIP trunk (voip provider or patton isdn).
I have a TE220BF available than i
On Tuesday 28 February 2012, upendra wrote:
hi,
an anyone tell me how to do auto dial to a SIP in Asterisk using a script
.any example will be more helpful ... !
Regards
Upendra
You need to inject a callfile into /var/spool/asterisk/outgoing . The file
should look something like
Just for fun I did something similar at one point.
0-9 A-D and * and # make a character set of 16 characters, perfect for encoding
as hex.
Take your string, get the ASCII value of each character, convert it to hex, and
add it to the encoded string.
Just before dialing, replace all e with # and
On 02/28/2012 09:06 AM, ml asterisk wrote:
Hi,
I have to make an asterisk gateway in front of several other asterisk.
This gateway will essentialy be used for outbound call.
This gateway will be connected to other asterisk by IAX trunk, outbound
call will use SIP trunk (voip provider or patton
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
I'd try turning off the jitterbuffer and see if that makes things better. I
just traced a similar call quality issue transferring calls incoming DAHDI on
one * box to another * box, and turning off the jitterbuffer on the side that
couldn't hear (in my case, the * box with the DAHDI lines, as
My first two guesses are that encryption is hosing you or that the
single-channel nature of IAX2 may have something to do with it. IAX2
talks on 1 channel, SIP uses twisted pair connotation on two channels
(as I understand it).
-Original Message-
From:
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
I've tried turning jitterbuffer off - doesn't make a difference. (And
why should it? The Jitterbuffer only applies to incoming calls, doesn't
it?)
On 2012-02-28 21:12:48 +, Noah Engelberth said:
I'd try turning off the jitterbuffer and see if that makes things
better. I just traced a
encryption=yes is meaningless if the provider doesn't support it (mine
doesn't). I put it there in the wild hope they eventually will - and no
config change will be needed on my part.
Still, when I changed it to encryption=no, and tested there wasn't any
difference.
So I've tried disabling
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
People around here either hate me or love me. I post experience and am
accused of bragging or whatever. As a reader, I want to know who is giving
me advice and what it is based on.
$40k/wk of long distance through VoicePulse. I have the invoices, that is
high usage, others attack me for
I have no interest in the penis-measurement competition firing up
here, but I'll say that we have 100% abandoned IAX from all of our
systems due to a myriad of issues. These days it offers no real
advantages in our opinion.
On Tue, Feb 28, 2012 at 4:03 PM, Steve Totaro
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
And the dude arrives talking about penis..
On Tue, Feb 28, 2012 at 6:07 PM, Carlos Alvarez car...@televolve.comwrote:
I have no interest in the penis-measurement competition firing up
here, but I'll say that we have 100% abandoned IAX from all of our
systems due to a myriad of issues.
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
An outside device can't register:
WARNING: getnameinfo(): ai_family not supported
WARNING: chan_sip.c:14456 parse_register_contact: Domain
'69.xxx.yyy.zzz:5060' disallowed by contact ACL (violating IP )
sip.conf:
[general]
...
alwaysreject=yes
dynamic_exclude_static = yes
allowguest=no
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
Hello!
I have problems with outbound faxes with asterisk 10.2 t38 gateway.
There is asterisk box, connected to panasonic kx-td500 over PRI link
with TE122.
If we try to send fax with following path:
panasonic 500 extension fax machine panasonic500- asterisk- ooh323-
cisco 3845- fax machine
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Imass
Sent: Tuesday, February 28, 2012 10:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Same provider
Eric thats really a nice idea to communicate between two or more of our
servers.
Make the call to the remote system and send the digits in the encoded
string, you will need something on the other end to decode the text.
But the other end is not our's but could be any solution which requires to
btw, played with res_fax.conf
if I set maxrate=7200 fax machines try (and fail) 9600 anyway.
Why? If limited ti 7200? looks like bug...
So I set maxrate=4800 and modems=v27.
Faxes pass
Looks like problems with V29...
29.02.2012 07:56, Dmitry Melekhov пишет:
Hello!
I have problems with
You want to allow single IP or whole subnet ?
Regards,
Zohair Raza
On Wed, Feb 29, 2012 at 4:44 AM, sean darcy seandar...@gmail.com wrote:
An outside device can't register:
WARNING: getnameinfo(): ai_family not supported
WARNING: chan_sip.c:14456 parse_register_contact: Domain
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