Displaying 6 results from an estimated 6 matches for "engleward".
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engelhard
2006 May 01
1
Using frequent keepalives to eliminate need for NAT port forwarding?
I have an asterisk system behind NAT, and need to
connect to public PSTN originators via SIP or IAX2,
but don't have the option of forwarding any ports
(4569, 5060, etc) to the asterisk system. However, the
NAT system does properly establish transient UDP
forwarding on the basis of outgoing connections, so is
it possible to configure asterisk to send frequent
keepalive UDP packets (say every
2006 May 01
1
/var/spool/asterisk/outgoing/ prematurely hangingup
Just a shot in the dark... but have you tried Answer() before
Playback()?
Josh McAllister
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom
Engleward
Sent: Monday, May 01, 2006 11:43 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] /var/spool/asterisk/outgoing/ prematurely
hangingup
I have a PSTN termination provider "foo" which will
accept standard U.S. calls in the form 1<10 digit
ph#>.
I have an outbound route...
2006 May 04
4
why a perfectly fine iax2 host becomes UNREACHABLE?
I've got this low-ping 100%-up dsl connection between two asterisk
1.2.7.1 servers. And oftentimes one of them would declare its opposite
UNREACHABLE.
Why can this happen? The host stanzas in iax.conf have raw IP's, so no
DNS monkey business here.. An inquiring mind wants to know.
2006 May 01
1
Using frequent keepalives to eliminate need forNAT port forwarding?
...PAT the port could be anything over 1024, but usually much higher,
and the originator will send to port 5060, which your NAT router will
drop.
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Tom Engleward
> Sent: Monday, May 01, 2006 6:25 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Using frequent keepalives to eliminate need
> forNAT port forwarding?
>
> I have an asterisk system behind NAT, and need to
> connect to public PSTN originators via SIP or IA...
2006 Apr 30
1
integrated voip originator, to digitize audio once and only once?
Calling 7777 from a local extension on my local
network, I get good voice quality from asterisk, and
asterisk reliably recognizes my dtmf input.
I set up a sipphone trunk (free) and called in to it
via a separate sipphone account on another computer,
and got slightly lower, but still good, audio quality.
I set up a FWD trunk (free) and called in from the
other computer, and got somewhat lower
2006 May 01
1
/var/spool/asterisk/outgoing/ prematurely hanging up
I have a PSTN termination provider "foo" which will
accept standard U.S. calls in the form 1<10 digit
ph#>.
I have an outbound route named "foo", with dial
pattern "5|.", with the only entry in trunk sequence
being "IAX2/foo".
I have an X-lite local extension, on which I can dial
51<10 digit ph#>, and asterisk will call out over foo
and the