I lost that when I re-arranged the configuration, but the problem to
start with was that the name between brackets in IAX.conf was not the
same as the username, even if I had a user= line in there.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On
Jared Davies wrote:
Is there any way to make the X100P to check for a dialtone before
dialing? My roommate is tired of hearing random dialing in his ear
when he's on the phone (same circuit as the X100P).
I know one solution is to just put in a TDM400P for the house, but I'd
rather get away
Hi,
Has anyone managed to get www.pcphoneline.com USB devices (FXS and Phone)
devices working with *.
It registers and works with an ease with FWD, but with *, it does not even
try to register.
I tried all sort of configuration settings and the PC I am on is using a
public IP.
It does not work
anyone out there got * to work with
a voicetronix openline4 card ???
Regards Mick
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi again,
When I run modprobe zaptel I get the message that the zaptel.o was
compiled for kernel version 2.4.20-4GB while this kernel version is
2.4.20-4GB-athlon. And fails.
When I run modprobe wcfxo I get the message that the zaptel.o was
compiled for kernel version
Yes, I have.
Ahh, lets see:
- I grabbed the redhat kernel src rpm, and recompiled.
- I then setup the vpb driver.
- Oh, a big thing. when you get the cards, MAKE SURE the revision is greater
than 19, otherwise it doesn't detect line drops correctly.
- And uhm. lets see. I had problems getting
Andrew
I am having trouble with
Sound ( only if you dialling from outside )
Cisco phone can not dial out
If I phone in and select extension number of Cisco phone
* dies
Any ideas ???
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Thorsten Lockert wrote:
And, given that a context= entry is not used for a peer, only for a
user, it also goes to say that if you use a friend, context is still only used
for the *inbound* portion of it. So I just don't really see why you so
strongly recommend against friend entries as opposed
On Sat, Oct 11, 2003 at 10:25:37PM +0930, [EMAIL PROTECTED] wrote:
Andrew
I am having trouble with
Sound ( only if you dialling from outside )
Hmm. not really. uhm. could play with the volume levels. or it could possibly
be something else.
Cisco phone can not dial out
Sounds more like
Title: Message
We
need an consultant
to
set-up a single line server as demo for our customers
needs
to run
voicetronix openline4 card
Cisco
7940 phones
please
email me off line
Regards
Mick
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 10 October 2003 19:10, WipeOut wrote:
gsm: 52 kbps (13 kpbs)
alaw: 154 kbps (?)
speex: 57 kpbs (24 kpbs)
These don't look right..
This is the setup
SIP--(nic1)Asterisk1(nic2)==(nic1)Asterisk2(zap)--e1pri/telco
The measured values came
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 10 October 2003 19:15, Eric Wieling wrote:
Check the mailing list. There are several messages addressing the
issue. The main cause of the extra bandwidth usage is IP and UDP
headers. VoIP sends small packets and so, many times, the
On Sat, Oct 11, 2003 at 10:45:26PM +0930, [EMAIL PROTECTED] wrote:
Sound is OK on inside phone
When calling in sounds bad
I don't think I ever got that happening - wasn't part of the scope, so to
speak. How do you mean sounds bad? choppy? it might be converting to/from a
high cost codec.
The PRI goes right into the T100P.
I forgot if the T100P is the one with more than 1xT1, if so, the 2nd T1 in
the T100P goes to the Channel Bank.
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of PBX
Sent: Friday, October 10, 2003 11:07 PM
To: [EMAIL
Hi
what can be wrong with * that console does not show any stderr text printed
from agi script?
I am starting with asterisk -rc
VERBOSE command does show text on console but printing of STDERR does not
I tried it from Perl and from Java and in both cases almost the same
Try putting at the top of the Perl script:
$| = 1;
select((select(STDERR),$| = 1)[0]);
This removes buffering.
quote who=Serge Mankovski
Hi
what can be wrong with * that console does not show any stderr text
printed
from agi script?
I am starting with asterisk -rc
Quoting Anton Tinchev [EMAIL PROTECTED]:
What gatekeeper do you use.
It seems that is programed to make outgoing calls only to registered h.323
users.
Just program it to forward unknown number to the asteris (or switch
everything to SIP)
I can't forward unknown number to the asterisk
Citeren Serge Mankovski [EMAIL PROTECTED]:
Hi
what can be wrong with * that console does not show any stderr text printed
from agi script?
I am starting with asterisk -rc
VERBOSE command does show text on console but printing of STDERR does not
If you are running
Hi all,
--
I tried to use * over satellite, but all my effort did not succeed.
The Asterisk is behind the VSAT and is resposibel for alle the SIP
clients in a field location.
The clients are notebooks and PDA's running SJPhoen for Windows and
PocketPC. Unfortunately
I could not find any
Thank you for your reply. I tried it, but it did not help
Serge
From: Robert Hajime Lanning [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problems with AGI scripts in Perl and Java
Date: Sat, 11 Oct 2003 10:37:33 -0700 (PDT)
Try putting at the
Which satellite system?
I think you need some specialized support, even special hardware. Check
out
http://www.groundcontrol.com/igvoip_001.htm
You may need to replace TCP/IP
http://www.mentat.com/skyx/skyx-gateway.html
Paul
Paul Mahler
[EMAIL PROTECTED]
phone: 650-207-9855
fax:
On Saturday, Oct 11, 2003, at 17:04 America/Chicago, Paul Mahler wrote:
Which satellite system?
I think you need some specialized support, even special hardware.
Check
out
http://www.groundcontrol.com/igvoip_001.htm
You may need to replace TCP/IP
http://www.mentat.com/skyx/skyx-gateway.html
I was looking into using satelitte for a backup internet connection at one
stage, iirc, its:
- 500ms transmit/recieve latency
- if yours sat connection terminates in the us, you should be able to reach
most place in 30ms
- if you're going to europe (from the termination of the sats in the .us),
Well, it's that stack that needs to go. Check out the link.
Paul
Paul Mahler
[EMAIL PROTECTED]
phone: 650-207-9855
fax: 877-408-0105
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: Saturday, October 11, 2003 4:03 PM
To: [EMAIL
Hi,
Iam successful in getting incoming calls for
my linejack
But, I am getting an error
cannot create channel of type
Phone
Does linejack do outgoing calls in asterisk
?
Thanks,PTA
I didn't get outgoing calls to work on the linejack cards. Which fortuately,
didn't matter. There where some notes outthere by people who said they got
it working. *shrug* didn't work for me.
If you use GNUGK, iirc, it works fine.
Thanks,
Andrew Griffiths
On Sat, Oct 11, 2003 at 08:24:59PM
On Sat, Oct 11, 2003 at 05:16:11PM -0700, [EMAIL PROTECTED] wrote:
I didn't get outgoing calls to work on the linejack cards. Which fortuately,
didn't matter. There where some notes outthere by people who said they got
it working. *shrug* didn't work for me.
If you use GNUGK, iirc, it works
So...
I would need as you noted two T100P cards or a T400P. The T1 goes into
the * Server and the second port of a T400P goes back to the asterisk
server. Then the extensions get broken out from the Channel bank?
Geoff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Exactly.
So...
I would need as you noted two T100P cards or a T400P. The T1 goes into
the * Server and the second port of a T400P goes back to the asterisk
server. Then the extensions get broken out from the Channel bank?
Geoff
___
I'm trying to create several contexts for extentions with
different levels of access to features and I'm wondering
how the heck do I include all the contexts so that you
can call internal to any extention in another context without
giving the features of the higher level context to the lower
level
I have a question regarding SIP and XTEN softphone.
Server IP - 192.168.1.102
sip.conf --
[7300]
type=friend
host=192.168.1.100
secret=1234
extension.conf --
; Test Sip
exten = 7300,1,Dial,SIP/7300
The sip phones registers, but I am unable to make any calls... Going
either direction... Any
[post re-ordered chronologically]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: Saturday, October 11, 2003 4:03 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP / IAX over satellite
On Saturday, Oct 11, 2003, at 17:04
Date: Sat, 11 Oct 2003 22:07:49 -0700
To: [EMAIL PROTECTED]
From: John Todd [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] SIP / IAX over satellite
[post re-ordered chronologically]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent:
Includes are recursive
Make a context with just all the internal extensions, and then make
contexts for all the outbound calls and another group of contexts just
as you are doing (admin, sales, etc)
Then
[admin]
include = international
include = extensions
[sales]
include = longdistance
Either way will work. Getting the T400 four port card gives you room to
grow, but getting 2 T100P single port cards saves you about $500.
Is this the only way to handle extensions... This turns a 4 port T1 card
into a 2 port card... Is this the suggested method?
Geoff
Hello,
unfortunately you can't do outgoing calls with LineJACK. If you have to
place outgoing calls, then buy some FXO VoIP Gateway (Micronet,
Audiocodes) or digium hardware.
Lubo
[EMAIL PROTECTED] wrote:
I didn't get outgoing calls to work on the linejack cards. Which fortuately,
didn't
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