I've searched the archives a bit and have not really come up with
a good answer to my queries.
I have * running on a RH9 box behind a LinkSys NAT box. I can talk
with iConnectHere outbound just fine. I am trying to configure an
inbound Xten softphone from outside. I have that user set as NAT in
Hi,
I need to know from your experience which is the minimum hardware
configuration (proc./memory) to run * with two X100P cards and around 10
internal IP phones (from which 4 are hardware ones and use G.711, the rest
of them with GSM), with a couple of IAX connections with other Asterisk
PBX'es
Hi,
What's the perfered way to launch Asterisk on boot, on say Redhat?
I tried putting this in my rc.local:
modprobe wcfxo
/usr/bin/screen -d -m /usr/sbin/asterisk -vvvcn
But it doesn't work, and I get this in my asterisk message log:
Aug 6 10:24:27 WARNING[16384]: File codec_g729b.c, Line
Hi Todd,
This limit on outbound calls looks interesting. Can you provide an example?
I have not used db routines before.
Thanks,
Ricardo Villa
http://www.telesip.net
- Original Message -
From: John Todd [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, August 07, 2003 2:51 PM
Hi All,
I recently purchased the Asterisk Developer's Kit (TDM) to try out
Asterisk. After following the directions in the Digium's FAQ topic entitled
Q. How do I configure my TDM40B and X100P?, I'm receiving the following
error:
WARNING[1074428608]: File chan_zap.c, Line 6748 (load_module):
I have already installed a small PBX ( 1 FXO (E100P) and 4 FXS
(TDM400P) )
but now i want to know how to build a bigger one... maybe 8 FXO and 24 FXS
something like that o bigger. But i dont know witch hardware i need.
use a
-1 t410 4 t1 spans
-32 ports off a channel bank(s)
-1 CAC adit 600
Hi Martin,
On Fri, 2003-08-08 at 00:16, Martin Stubbs wrote:
Unfortunately the present x100p driver code will not decode the callerid for 2
reasons
1) the UK protocol is different to the US system.
I have downloaded the specs and coding it would not be too difficult.
Maybe you could open
On Wed, 6 Aug 2003, Steven J. Sobol wrote:
and I found a reference to an AgentLogin.rtf. Looks great, except I can't
get it to work.
Mysteriously enough, now music on hold AND queues work.
Well, MOH works. Queueing works, almost. I can dial in from outside
and be placed in queue. If I log
the values for adapp and adsec as they are in CVS. As
far as I can tell those changes have no relation to problem this patch
solves, they're just local changes to satisfy your local preferences,
right?
actually those values (adapp and adsec ) should be parameterized in
adsi.conf
adapp - is
I'm not quite sure where to hunt for the answer to this. My apologies
in advance if this isn't the appropriate forum.
I have generally speaking used Kewlstart signaling on the POTS lines
that I have used with asterisk, with great success.
I have an installation where the POTS is coming from
On Thu, 2003-08-07 at 19:55, Wade Weppler wrote:
Hi Matt! :)
You can use the Local channel driver:
exten = _9NXXNXX,1,Dial(Local/${EXTEN:[EMAIL PROTECTED])
Where ${CONTEXT} is set to the local context you want to use.
What would be wrong with just a simple Goto?
exten =
To give you an idea of how interesting this would be: I could have a
peer with your office A, and if you had the appropriate filters on
your announcements to me, I would know how to reach office B without
going through office A or a PSTN connection. If you added a line in
office B, it would
HI!
Thanks for the Reply.
Servers and phones are on different VLAns. If it
was NAT problem, why does it work when rebooted and get failed( calls still
establishes but no sound ) after several hours. My codec setting in the phone is
G.711u which I hope is supported by *.
There was a thread a few months ago that tossed around some ideas for
using a cisco phone for intercom or paging. I don't have any ip
phones, and wondered if anyone had any luck getting intercom or paging
to work on the cisco units.
Do any of the (cheaper) ip phones have a way to support
Just got my new Budgettone phone, and I've got a couple of issues.
Most important, it doesn't seem to be querying for the time via NTP. I
put a sniffer on the line, and once it boots up the only outbound
traffic it generates is an attempt to contact a TFTP server, which is
programmed in
What phones are you using?
In my recent new Asterisk installation I'm having users complain that if
they answer a call on call waiting while talking on an existing line they
are then unable to park a call without one of the two parties hanging up.
Is there anyway whatsoever to be on a
Hi Andy,
I have a GE Dual CallerID cordless phone (analog) who correctly display the
name and the number when connected to an ATA with the default settings, so
it can pass the information to the analog phone connected to it.
It seems that we have here a minimum of two standards for CallerID on
Knowing very little about Basic Rate ISDN and having spent the last
couple of hours educating myself, I thought I would seek some more
informed comment. Please go easy if this is blindingly obvious :)
I have a ZyXEL Prestige 100 ISDN Router, a stand alone relic from when
we used to access the
Title: Message
I may
be entering this thread a little late, but just in case I'm not, the Snom 100
and 200 both support auto-answer very nicely.
Jerry
G.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
cwitteSent: Friday, August 08,
Not with your phones, unless they have auto-answer (and probably not
even then.)
I have heard of people with phones that support pager features on
the other pair of wires in an RJ-11. In other words, the second pair
of wires into the phone carries an audio stream that is played out
the
On Thu, 2003-08-07 at 17:10, Brian Capouch wrote:
The extension goes busy (from the perspective of someone else on the
PBX trying to call it) as soon as I plug it into the X100P, and when I
have asterisk pick it up I hear a faint busy signal.
You want to plug an ANALOG extension from your
However, when I load the driver (modprobe wct4xxp) for this card it
detects the card and loads the driver but then I keep getting all these
warning messages, such as:
Tried to load into 0004, but got instead
Not prepped yet
Every few seconds a similar warning message
A little off topic, but...
If anyone has a favorite url that lists model/types of sip phones, would
you send me an email at [EMAIL PROTECTED]
Looking to pick up a couple different varieties (including the Cisco 7960G
or equiv) for use in a lab test environment with asterisk.
Also, any
Seeing that many people here hit problems with activating their G.729
licenses, I decided to post my opinion.
I have purchased two G.729 licenses, for my private use. I did this even
though VoiceAge makes G.729 free for private use, as Windows
libraries. I guess a sufficiently motivated person
Michael Robertson, founder of both MP3.com and Lindows, has launched a
new company to supply inexpensive SIP phones ($129 for two) and related
services. See today's press release at
http://www.sipphone.com/tiki-index.php?page=SIPphone%20Inc
My question for the list is who will be the first to
thanks for the tip...
i already downloaded and compiled the required versions.
now i'm having a couple of errors while compiling h323 from
asterisk/channels/h323/
this only happend today from cvs, here is the error
cc -g -c -o
chan_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DNDEBUG
you use port source port + ip address and dest port + ip address.
IAX allows you to mux multiple calls in the same UDP data stream (called
trunking)
On Fri, 2003-08-08 at 15:26, William Flanagan wrote:
Eric-
Thanks for answering. Maybe a Stupid Question. If you have two IAX
softphones
Hmm.. then i don't know.. If you were using the GS IP phones I may have had an idea..
sorry.
ATT 957 analog sets
AJ
On Fri, 8 Aug 2003, WipeOut . wrote:
What phones are you using?
In my recent new Asterisk installation I'm having users complain that if
they answer a call on
struggling with localization issues (so the script is not German only)
took me a week longer than expected. (Did anybody ever get PHP's gettext
extension working??)
But finally, I've wrapped something up:
Hi,
I just tried to use it with our 7960 (sip-version).
I've set the services_url in
Hello everybody,
I have managed to setup an asterisk system with one hfc based isdn bri
card. In addition i use a cisco ubr 924 as an h323 gateway, and the
gnugk gatekeeper. I can call from h323 terminals and connect to the demo
context. I can also call any h323 terminal. Incoming calls
BTW, Digium needs a disclaimer for your patch. It was asked for here on
this list, and I'm not sure if you've reacted allready, but I'd really
appreciate it I you could send the disclaimer in. I'm working on
completing the options menu now, but it is not possible for me to
disclaim code that
Hi,
X-Lite logs into * with no problems. I dial 1000 and * plays greeting, but
i can not hear it.
Tried many times with the same result.
After quite few tries * complains about:
-
WARNING[81926]: File chan_sip.c, Line 388 (retrans_pkt): Maximum
yes it does.
Luciano
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Brian West
Enviado el: Martes 5 de Agosto del 2003 00:03
Para: [EMAIL PROTECTED]
Asunto: RE: [Asterisk-Users] GSM codec
Does your IOS on the AS5300 support g711ulaw?
bkw
On Mon, 4 Aug
Rattana BIV wrote:
The Caller ID is correctly passed when I receive a call from Phones.
But when Netmeeting call asterisk I only have the name of the channel like
H323:26022.
When I do in asterisk CLI the command : show channel H323:26022. I have next
Caller ID : (N/A)
That means that the
exten = _9X.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
regards
Martin
On Thu, 7 Aug 2003, David Hindmarsh wrote:
Hi
Is it possible to use a sip gateway as a trunk.
If so, how would I do this
David Hindmarsh
- Original Message -
From: Jamie Carl [EMAIL PROTECTED]
To: [EMAIL
I tried putting in
txgain=100%
rxgain=100%
and zaptel wouldn't load telling me I had wrong parameters in my zaptel.conf
i'll try again with txgain=5.0 but my setup is at a client so each time a
day passes
and i have to go round to the client just to try things out ... it's a bit
annoying!
my 2c
Hi,
Now that I have these QuickNet cards working (I have enabled ISA PnP and
voila`!), I need to set up VoIP to PSTN gatewaying... And what I need to
ask you is whether I am able to do this with one QuickNet LineJack card
or not?..
Another question: is there any good documentation (any
37 matches
Mail list logo
