Displaying 20 results from an estimated 41 matches for "abouttotri".
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abouttotry
2003 Jun 13
1
strace shows that files are not accessed
strace on file access in asterisk shows that * is not even attempting to
access the voice files.
If I *manually* load app_playback.so, app_macro.so, and then
pbx_config.so, I they will load and I get a dialplan. Ok, that's a
problem -- autoconf is clearly not working, or there's some other
related issue.
So I try to use the demo and do "dial 500". This should connect and
2006 Jun 09
0
Bad call quality using a certain channel.
Hi,
I am fairly new at working with Asterisk.
I am having a call quality issue that I really need to get ironed out before
we go to rollout the system in a week.
Any help would be greatly appreciated!!! Even if it is just pointing me in
the right direction.
My current setup:
I have Asterisk Setup on a Dell Server. It has 2 T100P cards. One will be
for out T1 PRI from the Phone Company (We
2006 Feb 11
2
No Voice when canreinvite=no
Hi all
I am using Asterisk 1.2.2 on frdora core 4. i have two
sip UA. if i put canreinvite=yes voice Ok on both
sides. and if i change canreinvite=no there is no
voice (media through asterisk)
one thing more if i try to use playback application
for playing some sound file it is also working (like
exten => 500,1,Playback(demo-abouttotry) this is
working).
here is sip.conf
2003 Jul 31
1
Zaptel cards, working FXS and SIP, no audio?
Greetings again all,
With the help of another list member I was able to get the my TDM400P
card working properly (and the PRI card loaded too for channels 1-24).
2003 Dec 16
1
asterisk and cisco call manager via h.323
Does asterisk work with CCM as gateway ?
When I trying call asterisk,I totally can't hear any sound.
When call ohphone - works good.
10.0.1.219 is CCM, 10.0.1.207 asterisk.
Trace messages here :
--------------------
== New H.323 Connection created.
-- Received SETUP message...
== Setting up Call
-- Calling party name: [5001,]
-- Calling party number: [5001]
-- Called party
2005 Oct 13
1
Noob help with IAX
Ok so I've just built and installed a CVS (HEAD) version of asterisk
on RHFC2 running a 2.6.13.3 kernel.org kernel. I installed the samples
via "make samples". Everything seems to work except one thing. I'm
trying to do the connect to the Digium IAX demo server portion of the
demo (dial 500) and I just get the following messages. I am behind a
NAT server and did NOT change
2003 Nov 28
1
Problem with SIP-Phones and * audio-files
Hi All,
I am a newbie to asterisk, and here is my first problem, where I do not
know any further.
I have to grandstream BT100 connected to asterisk. Working fine, for
calling to each other, and to call via a IAX-Link to the outside.
If I try to call the initial demo from the samples.extensions.conf I
have nothing to hear.
The CLI fine reports:
-- Executing
2003 Sep 20
4
Maximum retries exceeded w/SIP
First of all, I'd like to send a big "thank you" to all the folks who have
helped me get this far.
Now on to the next problem. Here's my current network setup:
The Big I ---+--- FreeBSD FW --- * (10.0.0.253) ---- PC (10.0.0.1)
|
+--- Laptop (public IP)
natd is set up with the following rules:
redirect_port udp 10.0.0.253:10000-20000 10000-20000
2006 Oct 21
2
1.4 branch on OSX?
I tried to update and build 1.4 (SVN-branch-1.4-r45775) tonight.
I took the additional step of nuking my modules directory first...
When I used the command asterisk -v to start asterisk, it seemed to go
along and get to the point where asterisk is running(ie Asterisk Ready).
At that point it was eating all available CPU.
I went ahead and tried to register a softphone to it via IAX2, which
2003 Dec 02
3
maximum retries exceeded
Hi,
i've just got 2 grandstream phones and when I try to connect them with *
I get the following:
-- Playing 'demo-abouttotry' (language 'en')
WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call 02da8d3d-a8de-c7e1-5f07-c032ab23d909@10.1.1.179 for
seqno 59134 (Response)
I've seen there was some discussion on this already but i
2004 Sep 29
1
Zaptal and Fedora Core 2 and losing GSM playback
Hi,
I've successfully installed Asterisk 1.0 on Fedora Core 2 with the 2.6.8
kernel. I have two other computers running X-lite connecting to it. I've
been able to set them up so I can dial extensions "123" and "124" to
talk between them. I'm able to access the default "1000", "500", and
"600" extensions and they all seem to work.
2003 Jun 27
2
Making calls from snom 100
Hello,
I`m trying to make a call from the snom 100( SIP mode) but whatever
number I dial I get a 404 error from Asterisk. Here are my configs and a
dump from "sip debug" . But if I make a call from a Zap line (see
extension 2382031), it rings the snom phone
sip.conf:
------------------------------------------------------------------------------
;
; SIP Configuration for Asterisk
2003 Dec 15
2
Beginners Question
Hi all,
New user to asterisk having just got it compiled and installed.
Running with no digium hardware (yet) and no soundcard in asterisk box.
Problem is using the sample configs with a sip phone added as follows
[2203]
type=friend
username=2203
secret=2203
host=dynamic
defaultip=192.168.0.2
dtmfmode=inband
canreinvite=yes
the console on * when running with -vvvvc says :- (whenb trying to
2004 May 04
1
Asterisk and windows h.323 gatekeeper calling problems...
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi there, i have a working Microsoft ISA firewall with buildin H.323
Gatekeeper....
So Far, i got registerd the asterisk on the M$ Gatekeeper...
here is the h.323 configuration:
; Open H.323 driver configuration
;
[general]
port = 1720
bindaddr = 0.0.0.0
allow=all ; turns on all installed codecs
dtmfmode=rfc2833
gatekeeper =
2005 Sep 05
6
asterisk CAPI dial-in issues
Hello configuration as follows, dial-out works:
capi.conf:
[hfcpci]
;;PointToPoint (55512-0)
isdnmode=MSN
incomingmsn=*
;msn=61
controller=1
devices=2
context=incoming
extensions.conf:
[incoming]
exten => _XX,1,Playback(demo-abouttotry)
exten => _XX,n,Dial,SIP/xlite1
exten => _XX,n,HangUp
When call is placed, the following debug info is shown, after the last
line, it stalls until
2006 Jan 19
1
Sound issue with Asterisk
Hey Steve and everyone,
I looked at the configuration, and unless I am missing something I don't
think they are configured
# ztcfg -vv
Zaptel Configuration
======================
Channel map:
0 channels configured.
In the zapata.conf file, it is the sample version, but I didn't notice
anything in there that related to what you said. Or is it in a
different file or location?
I am
2003 Jun 13
5
Applications, dialplan not loading
I've built the latest CVS of asterisk -- not the zaptel or libpri
directories, just the asterisk directory. asterisk installs
successfully, but there are severe problems. I built this system in the
past and ran it, but now building it again fails. This is the CVS as of
this morning, 2003-06-13, but I had problems on 06-11/12 as well.
After make; make install; make samples; make config, I
2004 Nov 26
2
Execute a script upon registration
Is it possible to execute a script upon successful registration and
authentication of a SIP device in Asterisk? For instance, have a script log
all successful registrations in a database or authenticate users instead of
using the secret=password in the sip.conf file? Thanks -
--
Brian Wilkins
Software Engineer
brian@hcc.net
Heritage Communications Corporation
Melbourne, FL USA
2004 Jun 21
0
dialplan help!-RESOLVED
All,
I was a bit too focused on where I thought the problem was - turns out
I wasn't crazy and the dialplan does work as expected. The problem was
with dtmf detection - setting relaxdtmf=yes did the trick. Sorry for
the premature post for help.
Begin forwarded message:
> From: Ben Witso <benw@bgwcomp.com>
> Date: Mon Jun 21, 2004 7:28:42 PM US/Central
> To: Asterisk-Users
2004 Dec 03
5
SIP SECURITY WARNING: v1-0 (cvs today) sip context in general section ignored goes to default instead - allowing unauthorized sip devices to place calls in default context
SIP SECURITY WARNING
Version: v1-0 (cvs today)
Problem: sip context in general section ignored - goes to default -
allowing unauthorized sip devices to place calls in default context
Fix [workaround]:
Remove or rename "default" context in extensions.conf
Notes:
I am not sure what other asterisk functionality may be affected by this
- review your other config