Displaying 20 results from an estimated 1000 matches similar to: "Setting variable for a context for all extensions?"
2004 Sep 15
4
IAX to IAX connect question
Hi,
I got my * working fine with FWD at office with 2 extensions, i receive
calls and i can make calls thru FWD. I got also my * at home, and i
connected it using auth=rsa. From my home, i can make calls using my office
iax, but if i try to redirect incomming calls from FWD to my * at home, it
rejects the call. I created the pub/key pairs for rsa and its working ok
and i just pasted the
2003 Oct 20
1
Setvar SIP_CODEC
Hello,
I have
a couple of 7960 and a quad T1 card on my asterisk box. I want to let
the phones to use g729 when they "talk" to each other, but to use g711
when I'm going to route the call out of my network using the T1 card.
Everything works just fine between the phones, but in order to be able
to make calls through T1 I have to disallow the g729.
For this purpose I have the
2005 May 28
1
Fax and SIP Device
A DID number was dedicated to receive fax, but i have the problem when
getting fax call,
which call will become a normal phone call and no fax was printed. When
fax is detected,
the fax extension is executed and dial the extension of the HT486 device
(firmware 1.0.5.22).
Somehow sending fax out working well. In the mailing lists, i notice
some are using HT286 and it work.
Could someone share
2004 Dec 01
0
extension and PSTN connection
I got two phones on an ATA-186 (601, 602) and two phones on the TDM22B
(603, 604). I have two lines on the TDM22B.
I cannot figure out some of the problems:
1. 601 dials via ZAP/3-1 to local phone number at PSTN:
ringing
pickup on PSTN (empty)
still ringing in the phone set 601
2. call from PSTN back:
601 picks up ... everything works !!!
No caller id shows up
3. For testing I have only one
2006 Feb 07
1
asterisk to FWD
Hello all,
Here is my problem,
I try to place a call to FWD (free world dialup) trough my asterisk PBX.
my config is as follow:
extensions.conf
----------------
[internal]
exten => 613,1,Dial(IAX2/iaxfwd-outbound/613) (service echo de FWD)
exten => xxxxxx,1,Dial(IAX2/iaxfwd-outbound/xxxxxx) mon numero FWD
exten => yyyyyy,1,Dial(IAX2/iaxfwd-outbound/yyyyyy) celui d'un ami FWD
2006 Feb 23
1
not consistent log from asterisk
Hello,
I have 2 channels in iax.conf
[iaxfwd]
type=user
callerid= Free World Dialup
inkeys=freeworlddialup
auth=rsa
context=incoming
qualify=yes
[iaxfwd-outbound]
type=peer
host=iax2.fwdnet.net
username=xxxxxx
secret=***********
auth=md5
The problem is:
When I tell FWD to call me I have this output in my asterisk
consol:
Executing Dial("IAX2/iaxfwd-outbound-3",
2004 Nov 30
0
No voice when I dial out
I can dial from 601 to a public number.
The public number rings. I pickup and hear nothing, while on 601 it keeps ringing.
(BTW, is it right to say "ringing" on the active phone?)
The *CLI> doesn't show me anything useful:
Executing Macro("SIP/601-8238", "dial-pstn|88097880|") in new stack
Executing SetGlobalVar("SIP/601-8238",
2004 Dec 13
1
incoming call from pstn to fxo not working with Asterisk
When somebody call me on my pstn # cable connected to my fxo card it does
not work when I check my computer the following error shows
Connected to Asterisk CVS-v1-0-12/05/04-19:46:25 currently running on
asterisk1 (pid = 2160)
Verbosity is atleast 3
-- Remote UNIX connection
-- Starting simple switch on 'Zap/1-1'
== Starting Zap/1-1 at incoming,s,1 failed so falling
2005 Jan 28
3
FWD and IAX2
Hi,
I had a FWD account set up with asterisk (using SIP) and it was working
fine both ways. I switched to IAX2 and now I can't get incoming calls
from FWD. People who call my FWD number get a "480 - user is not online"
message without any traffic reaching my box. I can call FWD numbers fine
over IAX2.
It seems fwd isn't trying to place the call over IAX2 because it thinks
2006 May 29
4
How to enable call waiting on Sip Phones
How do you enable call waiting on sip phones? Ive looked and googled and
can only find call waiting pstn phones butnot for sip. Is their a way of
setting this up within the dailplan?
2004 May 13
4
IAX Freeworld
I have looked all over the site(s) for help. But heres the problem. Im
missing something.
In coming works fine from FreeWorld via IAX. But when Dialing out i get:
May 13 13:42:01 WARNING[1150495040]: chan_iax2.c:5256 socket_read: I
don't know how to authenticate iaxtel to 65.39.205.121
my IAX.conf if as follows
[general]
port=5036
register => ######:xxxxxxxxxxxxx@iax2.fwdnet.net
2004 Aug 28
4
incomming call rejected using IAX2 with FWD
Hi,
I cannot seem to accept incoming calls from FWD using IAX2. I followed the
directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing
calls fine using IAX via FWD. When someone calls me from FWD I get the
following message:
Chan_iax2.c:5251 socket_read: Reject connect attempt from
65.39.205.121
Any ideas?
Thanks,
S.
2005 Jan 29
2
Call rejected by FWD: Unable to negotiate codec
When I try to call out to FWD over IAX2 I get:
Call rejected by 65.39.205.121: Unable to negotiate codec
I'm using asterisk-1.0.5 (the same settings works fine with *0.9)
I've standard settings in iax.conf
[general]
bindport=4569
register => xxxxx:xxxxxx@iax2.fwdnet.net
[iaxfwd]
type=user
context=fromiaxfwd
auth=rsa
inkeys=freeworlddialup
disallow=all
allow=ulaw
--
#Joseph
2005 Mar 15
1
Asterisk retains DTMF Control Even when an External IVR System is dialed
I am using Asterisk 1.06 Stable.
When I dial my Mobile Number to check Voice Mail or my Bank Account
Phone Access Number, the IVR System on the other end asks me to enter
*2378 to transfer to an attendant.
But When I press *2378, Asterisk tells me that it cannot transfer the
calls and gives an error on CLI saying Extension '' does not exist in
the dial plan.
What is the trick to make
2000 Jun 15
2
Checking the existence of a file
Is there a platform-independent way of checking in R whether a
given file exists in the user's filespace?
(so in a unix system, can you check within R whather, say,
/homef/jonm/thisfile
exists)
Thanks
Jonathan Myles
--
Dr. Jonathan Myles e-mail:jonathan.myles at mrc-bsu.cam.ac.uk
MRC Biostatistics Unit Tel. 01223 330372
Institute of Public Health FAX 01223 330388
2007 Jun 28
2
Linking Asterisk with another SIP PBX (or SIP Softswitch)
Hi List;
If I need to do a trunk between Asterisk and another
SIP softswitch (so Asterisk will send a SIP calls to
that softswitch), then I have to configure this on the
sip.conf file or where exactly? And is it the same
when I configure iax trunk?
Should I determine the context in this case for this
SIP trunk?
Regards
Bilal
2004 Sep 08
2
'connecting' voip-numbers to our Asterisk
Hi everyone!
I have a problem... We have received a couple of phone numbers for voip
from a local voip-provider. The work fine directly with a Cisco 7960,
but so far I've not been able yet to integrate them into Asterisk.
I've tried:
/etc/asterisk/extensions.conf
*****
[ip-incoming]
2007 Sep 12
2
Evaluating args in a function
Can anyone explain what I'm doing wrong here:
> fred <- data.frame()
> class(fred)
[1] "data.frame"
> test.fn <- function(x,class=class(x)) {class}
> test.fn(fred)
Error in test.fn(fred) : promise already under evaluation: recursive
default argument reference or earlier problems?
R 2.5.1 on both Windows and SUSE Linux.
--
Sanford Weisberg, sandy at
2006 Jul 11
1
Linux/MacOSX and "X11 protocol error: BadWindow..." warnings
This concerns behaviour which has been noted previously, for
example see
http://tolstoy.newcastle.edu.au/R/help/05/03/0844.html
http://tolstoy.newcastle.edu.au/R/help/03b/6873.html
and especially the thread
http://tolstoy.newcastle.edu.au/R/help/04/08/3025.html
I find that it affects both my Linux and Mac OS X setups.
It is a feature which has caused some (at least John Fox in
the
2006 Feb 16
2
Cisco 7960 won't register
Hello all, I've got a Cisco 7960 running version 7.4 firmware (heard there
were problems with 7.5) and I can't get it to register with Asterisk. I've
stripped down my configs on the phone to a bare minimum, and posted them
below. Basically, the Cisco phone sends absolutely no packets to the proxy
when it gets booted. If I make an outgoing call I see traffic getting to
Asterisk, but