Displaying 16 results from an estimated 16 matches for "zapchans".
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zapchan
2007 Nov 30
2
My AsteriskNo unable to registration
Dear The Expert,
I am very new with this, I have installed AsteriskNow, X-Lite as my
SoftPhone, I am using SPA-3102.
I have 3 extensions,
me at 250, 998 is my Linksys SPA-3102 and 999 for PSTN Line (see below)
My problem is, I am unable to call 998, I thought this is registration
problem, (because the Linksys screen info said Registration Failed)
Could any body please help?
Many thanks in
2007 Jun 04
2
FX Dialing Odd
...ternal
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
callgroup=1
pickupgroup=1
immediate=no
;define channels
context=internal
signalling=fxo_ks
channel => 1-2
context=zapchans
signalling=fxs_ks
group=1
channel => 3
extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no priorityjumping=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;TRUNK=Zap/g1 ; Trunk interface
TRUNK=Zap/3...
2007 Apr 17
2
peers are using wrong contexts
Hello, everyone.
Today I've installed an asterisk svn trunk (r61667). The problem I'm
having is no matter what context I set in the config file for that peer,
"default" is always being used.
The output of "sip show peers" shows the context correctly, but when I
try to make a call, using that peer, I can only dial the numbers set in
the "default" context.
2008 Jan 28
2
Dial agent channel - busy
Hi,
when I'm trying to call the following extension
exten => 6002,1,Verbose(1|Extension 6002)
exten => 6002,n,Dial(Agent/6002)
exten => 6002,n,Hangup()
the call is terminated and I get the following warning from asterisk:
app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Agent'
(cause 17 - User busy)
When calling the agent with Dial(SIP/6002) no problem
2009 Jan 16
0
No subject
---
span_1 = DAHDI/g11
1,1,dial(${span_1}/${EXTEN:0})
---
The configuration was rsync'd from a working pair of asterisk servers in
another office. The only difference was the version 1.4.22 for the original
servers that were operating as expected, 1.4.24 and 1.4.24.1 for the new
servers.
Included in both working and non working servers is the following
configuration:
2007 Apr 19
1
users.conf SIP registration fails
I recently upgraded from asterisk 1.2.13 to 1.4.2 and am looking at using
the users.conf file to setup my users, before i was using real time SIP
which worked fine. However when i create a user in users.conf i am unable to
register the user form a softphone, however that same softphone can still
register a different the users i currently have setup form the sip.conf from
real time. i've
2008 Apr 03
1
Hearing "transfer" during call
Hi list,
I enabled the transfer function in my dialplan, but I may configure it
wrong because sometime when I call a SIP extension number from one FXS
port, the SIP side will hear word "transfer", I hear nothing, after
that, the call conversation is fine.I'v had this problem for a long
time, could not get clue where I configure it wrong. here is my
related config part:
sip.conf:
2007 Sep 13
1
Problems with two trunks
Hi,
I am attempting to setup an asterisk server, current specs:
CentOS release 5 (Final)
Asterisk 1.4.11
Asterisk-gui checked out from SVN last week
I started with a fairly basic setup involving one VOIP provider who
provided one dial in number, and a couple of handsets. Config files are
below. It was pretty much totally built by Asterisk-gui, except for the
fact I had to add
2007 Sep 13
2
FW: Problems with two trunks
Update on this:
I found that by changing insecure = very to insecure = invite, adding
the second trunk no longer stopped calls working.
I've read the documentation on this switch and still don't see how it
applies/is meant to get used.
Anyway, with this change in place, the following may help:
asterisk*CLI> sip show registry
Host Username
2007 Mar 13
6
Asterisknow with video and X-Lite not quite working
Hello everyone,
I have previously asked this question on the asterisk-video list, but I
got directed here.
I have a setup consisting of asterisknow beta4 (not sure if that is
crucial) and a few clients all running X-Lite 3.0 (not eyebeam) on the
local network. My computer has a USB-Camera installed, and now I would
like to do some video calling with it, at least, so that the other user
can
2008 Feb 22
1
Message waiting light on polycom 301 using asterisk 1.4.14
All,
I am setting up asterisk on a nslu2 (Linksys) using unslug.
Everything is working great except that I have a polycom 301 and I
cannot get the message indicator to work. I have created the users and
mailbox in users.conf and I can manually dial the mailbox (*986000).
Last thing is I am not using config files for the polycom just web
browser.
Can anyone point me in the right direction
I
2007 May 17
5
DUNDi configuration problem
Hi peeps,
I've been struggling with DUNDi for a few days now and I can't seem to
make call from Asterisk A to Asterisk B. If I do a "dundi show peers",
it finds the other peer but I can't seem to make any calls. Can
anybody help me out here.
Here's the situation:
Machine 1: Debian with Asterisk 1.4.4 --> 192.168.1.103
Machine 2: AsteriskNOW --> 192.168.1.69
The
2007 Jul 12
0
No subject
Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D =
31999)
-- Remote UNIX connection
Verbosity is at least 8
-- Executing [00425298582 at numberplan-custom-1:1]
Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new
stack
-- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20",
"SIP/trunk_3/0425298582")
2007 Jul 12
0
No subject
Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D =
31999)
-- Remote UNIX connection
Verbosity is at least 8
-- Executing [00425298582 at numberplan-custom-1:1]
Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new
stack
-- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20",
"SIP/trunk_3/0425298582")
2007 Jul 12
0
No subject
Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D =
31999)
-- Remote UNIX connection
Verbosity is at least 8
-- Executing [00425298582 at numberplan-custom-1:1]
Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new
stack
-- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20",
"SIP/trunk_3/0425298582")
2007 Jul 12
0
No subject
Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D =
31999)
-- Remote UNIX connection
Verbosity is at least 8
-- Executing [00425298582 at numberplan-custom-1:1]
Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new
stack
-- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20",
"SIP/trunk_3/0425298582")