As I understand it, & if I understand you correctly, the register
parameter is for the client side. The nat=yes parameter is for the
server side, so it has nothing to do with your register statement. The
sip debug displays "no nat" because sip.broadvoice.com is not "behind"
the nat, it's "in fron
g729 must be licensed from voiceage, through digium. You can purchase
licenses from digium's online store @ digium.com.
-g
On Thu, 2004-06-17 at 16:37, Carlos Medina wrote:
> Hi there, im having some problems with my asterisk box, it seems codec
> is the principal cause of it. Reading in some f
Folks,
Randomly, when the phone is taken off-hook, the the Iaxy produces a
irritating banshee scream as opposed to a dial-tone. Cycling the power
fixes the issue, & sometimes it magically goes away by itself.
Has anyone experienced this issue & potentially fixed it?
I'm using asterisk CVS
I'm using this scenario & have had no problems.
When you say that HT-286-2 begets silence, do you mean it does not ring,
or that when you pick up after the ring there is no audio?
-g
On Fri, 2004-06-18 at 14:31, Nathan Martinez wrote:
> I have 2 Grandstream HT-286 devices and an Asterisk ser
-
I'm here with you on this one. I've not been able to figure this out -
I triple & quadruple checked that I have the right versions of pwlib &
openh323, & I've followed all recommendations in the README, yet I still
do not have audio in both directions.
I'm also using a cisco 5300, & there i
Sorry, Tom, I missed this message when it came through. It seems this
problem is a continuing issue among the asterisk folk.
Tell me, what versions of IOS have you tested with, & do you have any of
the h323 options enable/disabled in the 5300?
-g
On Fri, 2004-06-18 at 21:09, T. Chan wrote:
I have no new information, just a note of encouragement to those
traversing the bowels of h323:
I've been trying to get h323 working with asterisk for several months
now, trying with chan_h323 & chan_oh323 with all kinds of different
combinations.
As with several folk on the list, I've had no
I would like know if the asterisk handle each protocol
> (SIP and H323) separatedly or if the asterisk
> translate the protocol?!?!
Yes & yes.
Though getting h323 to work seems to vary from system to system.
-g
On Tue, 2004-07-06 at 14:34, Giscard Fernandes Faria wrote:
> Hi guys, I am a ne
Has anyone noticed a delay in sending DTMF to zaptel devices?
I have a T100P connecting to an NACT telecom switch. All calls are sent
just fine, but there are 4 seconds of delay between when the channel
goes off-hook & the digits are collected by the NACT switch.
I also have some cisco 5300 bo
By the way, I'm not using the "r" option - or any options for that
matter. Here is my what I'm testing with in extensions.conf:
exten => _.,1,Dial(ZAP/g1/${EXTEN})
g1 is a group to a T1 using e&m wink.
-g
On Tue, 2004-07-06 at 15:44, Glen Hinkle wrote:
>
I assume the pstn is your * system.
Can you get audio both ways if you send the traffic back to *?
pstn -> as5350 -> pstn ?
-g
On Thu, 2004-07-08 at 14:09, [EMAIL PROTECTED] wrote:
> Hi all.
>
> I have a strange problem, I've got a AS5350 hooked up to a telco using
> two trunked E1's
>
>
What's your relevant dial peer & sip.conf config?
-g
On Fri, 2004-07-09 at 03:49, Mikael Andersson wrote:
> Glen Hinkle wrote:
> > I assume the pstn is your * system.
> > Can you get audio both ways if you send the traffic back to *?
> >
> >
Are there any debugging tools for the digium zaptel cards that would
report the activity on the line, such as DTMF and/or connection
protocol?
I'm looking to debug the connection with a T100P, & I don't have $2000
for a T1 test set.
Thanks,
Glen
_
d by a "debug
Zap/1-1" as soon as the call is attempted. I get the following message
repeatedly until the call is connected:
"<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1]"
Any ideas or directions are appreciated.
-g
On Mon, 2004-07-12 at 19:14, C. Maj wrote:
&
> As I understand, oh323 channel cannot act as h323-gatekeeper and it
just role as a simple h323-GW ( am i right micheal).
Correct.
> If this is the case, how can I register my h323 EP ? shall I user a
3rd party GK like (gnu or mvts)?
Correct. Mvts is expensive, OpenH323gk is fantabulous,
The call is inbound on the pots dial-peer, so you should use incoming
called-number, as opposed to destination-pattern.
dial-peer voice 1 pots
incoming-called number [0-9]T
no digit-strip
direct-inward-dial
port 3/0:D
I'm not familiar with the [0-9] syntax, but if it works, ok. I usually u
I'm getting random Red Alarms with my T100P. I am using cvs zaptel from
July 13, & have tried using remove clock source & local clock source.
There doesn't seem to be any warning: every couple of days the line goes
down, & then around 5 seconds later, it comes back up.
If anyone as seen this beh
I'm getting red alarms with my T100P card that last from 4-15 seconds.
They seem to happen randomly, every couple of days.
Has anyone seen this behavior, or does anyone have any ideas regarding
what would be causing it?
Thanks,
Glen
___
Asterisk-
I'm getting this error continuously when sending to a cisco 5300:
frame.c:120 ast_smoother_feed: Dropping extra frame of G.729 since we
already have a VAD frame at the end
The connection is highly intermittent, sometimes there's a ring, other
times there is not. Is there a way to completely dis
Yes, presuming the 1U can accept any standard pci card.
-g
On Fri, 2004-07-23 at 18:06, Sathya wrote:
> Hi,
>
> Can a wildcard T100P be installed in a 1U server ??
>
> Sathya
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.c
This will explain the basics of * configuration:
http://www.digium.com/handbook-draft.pdf
After reading this document, you will understand contexts.
-g
On Tue, 2004-07-27 at 10:53, Thomas Kuepper wrote:
> Am 27.07.2004 um 15:37 schrieb Philipp von Klitzing:
>
> > Hi!
> >
> >> Now my quest
For anyone interested, the banshee screen I was experiencing was due to
my cordless phone. I used a normal corded phone without separate power
& it was fine.
I suppose there was some type of power overload that the iaxy couldn't
handle.
-g
On Tue, 2004-06-22 at 17:20, Andre Gironda wrote:
Just write a CGI script that places a file in in the outgoing calls
directory. /var/spool/asterisk/outgoing, I believe. This will
accomplish what you're wanting.
-g
On Mon, 2004-08-09 at 11:45, Andrew Thompson wrote:
> Andrei Goncalves wrote:
> > Hello !!
> >
> > I saw in FWD site a phone o
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