Hi,
Dotheg729 codec licensesfor
Asterisk workon aSIP environment (only SIP UAs running g729+
Asterisk)? I would liketo buy a couple for a SIP test labbut I
have not found any documentation on wether it works for SIP UAs or not.
The Digium page only mentions: "The G.729 codec works with all
Wondering, since they appear to be plentiful on eBay, whether I could
get a Nortel 350 to use to learn my way around ADSI.
The vendor claims that these are generic, and looking through the
archives I wonder if that means that they might be unlocked in the sense
that the word is meaningful to
Mark:
Will you add this to cvs?
On Sun, 2003-07-27 at 04:06, Andy Hester wrote:
Tilghman,
I applied your voicemail_prompts patch and it works like a charm. Thanks
for donating the code and thanks to those that donated the voice prompts!
Another win for Asterisk
Sincerely,
Andy
Hi,
I have asterisk behind my primary PBX connected via ISDN (chan_capi).
Calling out and calling in works just fine, however I can't connect to
my primary pbxs' extensions.
If anyone has an example extensions.conf, I'd be grateful for a copy.
I tried (the MSN of the ISDN card is set to 30)
Hey all,
As there seem to be some problems with DTMF-signalling between chan_sip
and several clients, due to which many could not properly dial a number
at the dial-tone of the XS4ALL-gateway at FWD-number 42442, I've now
arranged for a prefix on FWD for this gateway.
From FWD, you can now dial
What channel banks are best supported by asterisk, and available new at
preferably decent prices??
It would seem that for small offices with less than 15 users, a single port
T1 card with a channel bank, with say 15 FXS and 9 FXO (or similar config)
would be ideal. So I would like to find channel
So you mean a just simple blank line at the end of the musiconhold.conf
file or the extensions.conf file?
Second question, though it might seem a bit stupid, do I perhaps need a
sound card on the box that asterisk is running on? I don't think this
should be the case but I'm just wondering.
I put a TE410P card in a machine (a Tyan 2665 with 2x2.4GHz Xeons). A
red flashing light circles around the 4 RJ48C sockets. I load the
wct4xxp driver, and the flashing light stops. Whether I connect an E1
signal or not, no lights are shown, and no alarms are reports in the
/proc/zaptel/XXX
we have now perfect results with yesterdays cvs and the te410p
todays cvs allways thinks that immediate is set to yes in zapata.conf. weird
...
cheers
Michael
On Sunday 27 July 2003 7:12 pm, Mark Spencer wrote:
I put a TE410P card in a machine (a Tyan 2665 with 2x2.4GHz Xeons). A
red
OK Funny guy,
Mark Spencer wrote:
I put a TE410P card in a machine (a Tyan 2665 with 2x2.4GHz Xeons). A
red flashing light circles around the 4 RJ48C sockets. I load the
wct4xxp driver, and the flashing light stops. Whether I connect an E1
signal or not, no lights are shown, and no alarms are
Hi,
You can try to apply Michael's patch... for me it works perfect.
BR,
Dan
- Original Message -
From: Ricardo Villa [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, July 26, 2003 5:47 PM
Subject: [Asterisk-Users] PCM Voice Quality Issue on CVS Version
Hi,
I have
Hi,
I tried todays CVS and it works fine now.
Thanks,
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 27, 2003 12:40 PM
Subject: Re: [Asterisk-Users] PCM Voice Quality Issue on CVS Version
Hi,
You can try to apply Michael's patch... for me
Yes always end your conf files with blank lines otherwise you may get weird results
from asterisk..
as for the sond card requirement I don't know all my systems have onboard sound..
So you mean a just simple blank line at the end of the musiconhold.conf
file or the extensions.conf file?
OK, this is probably a dumb question for a lot of you, but I have no
experience with digital lines outside of a tiny bit of ISDN, so I'll
just bite the bullet and ask some newbie questions. I am attempting to
plan an asterisk installation with about 20 SIP phones and the following
incoming lines:
Ok I have festival on RH8. It speaks fast and you can't understand it. I
don't have any FXO cards in this box yet. Can someone shed some light on
this?
Thanks,
Brian
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Sun, 2003-07-27 at 13:29, John Laur wrote:
OK, this is probably a dumb question for a lot of you, but I have no
experience with digital lines outside of a tiny bit of ISDN, so I'll
just bite the bullet and ask some newbie questions. I am attempting to
plan an asterisk installation with
Title: Message
Has anyone got the
BT Speedway (AVM Fritz) card working on a RedHat 8.0 system with
*.
If so could someone
give me some pointers on getting the right sequence of installingthe
drivers and which versions to use.
Thanks,
Stuart
On Sunday, July 27, 2003 3:32 AM, Brian Capouch
[SMTP:[EMAIL PROTECTED] wrote:
Wondering, since they appear to be plentiful on eBay, whether I could
get a Nortel 350 to use to learn my way around ADSI.
The vendor claims that these are generic, and looking through the
archives I wonder if
Hi,
I am in the process of recording voicemail prompts in german. How do I
specify the language for the voice mail messages? I want to offer both
language files, based on the calling party.
Any and all help is greatly appreciated.
rgds
pos
___
Assuming it is a suitable Fritz card your best bet is to get the CAPI
library/driver from AVM and then check this out
http://www.junghanns.net/asterisk/ - chan_capi is reportedly the best
performing ISDN channel driver for asterisk, although I personally haven't
used it ;-)
Iain
--On
Use setlanguage. Then organize the language files by directory e.g.
/var/lib/asterisk/sounds/de
/var/lib/asterisk/sounds/digits/de
Also, say.c will have to be modified to support German style number
handling.
Mark
On Sun, 27 Jul 2003, Peer Oliver schmidt wrote:
Hi,
I am in the process of
I feel like a T1 with 24 channels should suffice, but what exactly
do
I order and what to I have to have in my asterisk unit to interface?
Does the line they terminate just plug into a T100P or do I need
some
extra hardware? What services do I need to be sure I order on the
T1?
Is there
I am working on one (chan_h323).
On Thu, Jul 24, 2003 at 07:50:08PM +0200, Peer Oliver schmidt wrote:
Is there a Debian package available for asterisk-oh323, or the chan_h323?
If yes, where might I find one?
Thanks
rgds
pos
___
Asterisk-Users
Hi Peer,
at my site it is working exactly as you wrote in your 1st example. How is
your PBX setup? I remember that there is a way to set a pbx to spontanic
trunk access. At least my Agfeo has got such a setup possibility. Try to
switch this off for your ISDN Card.
Cheers
Andreas
On Saturday 26 July 2003 21:06, Andy Hester wrote:
Tilghman,
I applied your voicemail_prompts patch and it works like a charm.
Thanks for donating the code and thanks to those that donated the
voice prompts! Another win for Asterisk
Is anybody at all using the variable substitution
I would also like to see a patch to ignore voicemail messages x number of
seconds long.. ususally those 1-4 second voicemails are nothing anyway..
bkw
On Sun, 27 Jul 2003, Tilghman Lesher wrote:
On Saturday 26 July 2003 21:06, Andy Hester wrote:
Tilghman,
I applied your
Tilghman,
I'm not sure how to use this logic. Would this be for something like, for
example, deleting of forwarding a message that a certain age?
Andy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tilghman
Lesher
Sent: Sunday, July 27, 2003
Title: Message
All you really should need is:
modprobe hisax type=27 protocol=2
id=isdn0
and in modem.conf:
driver=aopendriver=i4ltype=i4l; ISDN
example;group=1msn=xxxdevice = /dev/ttyI0device
= /dev/ttyI1
Has anyone got the
BT Speedway (AVM Fritz) card working on a RedHat 8.0
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