Hi,
if you turn off the reinvite in the asterisk configs for those ata186s
then it will pass through asterisk even if asterisk doesn't understand
the codec.
So I must have:
canreinvite = no
in sip.conf file for the specific phone?
Then the call is passed through Asterisk without any
On Sat, 2003-05-31 at 10:51, Dan wrote:
Hi,
if you turn off the reinvite in the asterisk configs for those ata186s
then it will pass through asterisk even if asterisk doesn't understand
the codec.
So I must have:
canreinvite = no
in sip.conf file for the specific phone?
yes
Then the
But I've got a stack of flyers and promo sheets that say $ISA_CARD work
fine shareing IRQs! /troll heh.
Nick
On Wed, May 28, 2003 at 05:24:02PM -0700, [EMAIL PROTECTED] wrote:
On Fri, 23 May 2003, Stephen R. Besch wrote:
own card to determine if it is the interrupting hardware.
In
There is one more note: make sure you don't have any options in your
Dial statement that require the Asterisk server to do transcoding.
Such options would be r, or m, or t, which will cause Asterisk
to need to listen and/or insert sounds in an audio stream if I
understand previous
I also have had very good luck with BigBrother, running it at home and
at several workplaces. It's very easy to extend, and quite reliable,
along with free for non-comercial.
Nick
On Sat, May 31, 2003 at 03:29:14PM +1000, Adam Goryachev wrote:
That brings to my mind a question...
Many thanks,
Dan
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 31, 2003 7:29 PM
Subject: Re: [Asterisk-Users] Passing audio stream through Asterisk or not?
On Sat, 2003-05-31 at 10:51, Dan wrote:
Hi,
if you turn off the
I have an adit 600, but Im not using the second span
I did have some small difficulties getting a single link to come up, had
to set * to provide timing to the line, and the CB to recover it from the
line
If you get it working, could you let us know what it was?
what was the exact
set
you are quite rude . . . why don`t you think tjat the time is passing on
and the new people and their obligations are waiting.
just a think
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Just wondering..
if there a way to specify a specific codec to use for a call between
two asterisk machines..
eg:
switch =
dial =
is there such an option we can use ?
.
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Can anyone tell me what I need to do to tell the voicemail and voicemail2
applications to download to a different slot? The only problem I have
remaining is this one... I dial the voicemailmain extension from a PT480,
and the display tells me services is full, download refused, but asterisk
PBX
After mucking around with it a bit, I was able to get it to sync with the
channel bank as the clock source for the line:
jsci-cbank1 show clock
Primary Master Transmit Clock Source:Internal
Secondary Transmit Clock Source: Slot A DS1 2
Also, are you using T1 crossover cables?
IAX peer's and user's can have specific allow and disallow directives.
Jeremy McNamara
Gary wrote:
Just wondering..
if there a way to specify a specific codec to use for a call between
two asterisk machines..
eg:
switch =
dial =
is there such an option we can use ?
.
A working example for restricting codec's by peer
[general]
port=5036
bind=0.0.0.0
allow=all
tos=lowdelay
[NuFone]
type=peer
disallow=all
allow=iLBC
trunk=yes
host=switch-1.nufone.net
context=NANPA
I have this iax.conf setup on many of the customer boxes I babysit. This
way they can use any
Is the remote ATA behind NAT ? Is it set to register SIP ?
On Sat, 31 May 2003 20:21:17 -0400, Jeremy McNamara wrote:
A working example for restricting codec's by peer
allow=iLBC
now thats is interesting
I haven't seen mention of iLBC on the list still trying to get
speex going here :-)
.
___
On Sat, 2003-05-31 at 21:34, TC wrote:
Yea looks like i have a issue with t400 spans 2 4 :(
I now can get 2 pretty green lights on span 1 and span 3
on the adit 600 on T1-1 and T1-2 ..and the 2 fxs/3fxo card are all green
So i plow'd ahead on possible shakey ground ...
Eep... Bad card
But now i dont get any dial tone when i pick up a hand set, the adit 600
recognizes the off hook goes amber but * does not see it off hook
And it has registered the channels
== Parsing '/etc/asterisk/zapata.conf': Found
-- Registered channel 1, FXO Kewlstart signalling
I think you may
Hi
Why if the Wildcard X100P is really a modem with chipset motorola
62802-52, why you can use another modem as the same form.
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its not only the modem chip involved.
Consider it a whole chipset issue :-)
On Sun, 1 Jun 2003 00:25:56 -0500, Jorge Cisneros Flores wrote:
Hi
Why if the Wildcard X100P is really a modem with chipset motorola
62802-52, why you can use another modem as the same form.
On Sat, 31 May 2003 20:21:17 -0400, Jeremy McNamara wrote:
A working example for restricting codec's by peer
allow=iLBC
now thats is interesting
I haven't seen mention of iLBC on the list still trying to get
speex going here :-)
.
[EMAIL PROTECTED] asterisk]# cd /usr/src/asterisk; du
fyi, i used ilbc through nufone today, much better than gsm, less
bandwidth usage (~20% less, ymmv), less cellular-syndrome and for some
reason iax seems to like it more too...
no call drops (did a 30 minute conversation) and mostly solves the quality
issues i've had with gsm, due to ilbc's
thanks Luke :-)
On Sat, 31 May 2003 23:17:34 -0700, John Todd wrote:
On Sat, 31 May 2003 20:21:17 -0400, Jeremy McNamara wrote:
A working example for restricting codec's by peer
allow=iLBC
now thats is interesting
I haven't seen mention of iLBC on the list still trying to get
speex
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