Displaying 20 results from an estimated 20000 matches similar to: "Installing Asterisk on Mandrake 10.1 Official"
2005 Jan 23
0
Upgrade to the newest cvs now asterisk will not start
Hello group
I just update to the newest CVS now I'm not able to get asterisk to
start. No error during the make or make install
I did a make clean before the make;make install
Any help would be great!!!!
Here is the output
asterisk -vvvvvgcd
Parsing /etc/asterisk/asterisk.conf
Parsing /etc/asterisk/extconfig.conf
== Binding realtime_ext to mysql/realtime/extensions_table
== Binding
2006 Mar 13
2
Unknown signalling method 'pri_cpe'
Anyone have any idea what this is talking about.
Here is my zapata.conf
[channels]
switchtype=5ess
signalling=pri_cpe
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
callerid=asreceived
group=1
context=default
musiconhold=default
faxdetect=incoming
channel => 1-23
Here is my zaptel.conf
span=1,1,0,esf,b8zs
bchan=1-23 # set this to 1-15,17-31 for E1
dchan=24 # set this to 16 for
2005 Jan 23
0
Upgrade to the newest cvs now asterisk will notstart
Looking at the error I tried moving chan_modem* out of the modules
folder and asterisk started and its working again...
Not sure what changed in the chan_modem_i4l.so but removing it from the
folder fixed my problem.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric Hall
Sent: Sunday, January 23, 2005
2004 Sep 14
1
asterisk does not start...
When I do a 'asterisk -vvvvvc' I get following, but asterisk does NOT stay up:
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
2005 Aug 11
2
wildcard/FXO config
Trying to config the latest Asterisk/zaptel with an Digium Wildcard and a
single X100m FXO interface connected to a POTS analog line.
Build and install of both work ok - I'm using Suse 8 on a dual Pentium box. I load
the driver with "modprobe wctdm" and the LED on the wildcard lights up. Then I start
Asterisk with "asterisk -vvvgc" and asterisk fails to start.
The
2003 Jun 29
1
SIP only with no soundcard?
Skipped content of type multipart/alternative-------------- next part --------------
[root@LINUXVM root]# asterisk -vvvc
== Parsing '/etc/asterisk/asterisk.conf': Found
Asterisk 0.4.0, Copyright (C) 1999-2001 Linux Support Services, Inc.
Written by Mark Spencer <markster@linux-support.net>
=========================================================================
== Parsing
2005 Feb 19
3
Still asterisk startup crash plz help
Hi,
First i would like to thank the kind people of the list who have
answered my previuos mail, but i am still stuck as asterisk still
crashes upon startup, i have read the install article at
http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation
and i have search the asterisk archives, but i still cant get asterisk
to work, i have tried reinstalling asterisk but it still complains and
2004 Jan 03
1
Newbie - getting two local phones tocommunicate would be a good start :)
Hi John,
Try adding username=5702 and username=5703 to each of the configs in
sip.conf. I recall I had this problem with the Grandstreams.
-----Original Message-----
From: John Coll [mailto:john.coll@csoft.co.uk]
Sent: Saturday, January 03, 2004 11:56 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Newbie - getting two local phones
tocommunicate would be a good start :)
2005 Oct 10
0
Asterisk behaving wierd!!
hello,
I have been using asterisk now for about 2 years now on a RH8.0 it is our
main call gateway.
I have on the box 3 T1 TDM cards connected to 2 Rhino channel
banks (FXS) and 1 CAC Access bank I (FXO) with so many softphones and ATA
186s.
It has been working good till today some few hours ago. i just
discovered that there were no dialtone on the phones.
Asterisk did not spit out any error, it
2005 Jun 15
1
app_dial.c:977 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)
Hi,
Ive been struggling with asterisk for a few days now. I understand
pretty much how it works and how to tie things together (SIP -> SIP
internally works fine etc), but just cannot get asterisk to work with an
X100P clone (its a Ambient MD3200, if that means anything to you guys).
I have tried (initially) asterisk 1.07 with zaptel 1.07, and now
Asterisk CVS-HEAD with zaptel cvs. Both give
2010 Feb 02
0
Issue when reloading
Hello list!
I?m having an issue when reloading Asterisk, I?ve had this problem in
Asterisk 1.6.1.6 so I upgrade to 1.6.2.1 version, but I still have the same
error.
For example, I send a "reload" in Asterisk CLI and this is the output:
isb152*CLI> reload
== Parsing '/etc/asterisk/extconfig.conf': == Found
== Parsing '/etc/asterisk/manager.conf': == Found
2003 Sep 17
1
core dump back trace of chan_oh323
hi michael,
here are the core dumps.
only kphone works when 0.5.5 and * cvs.
audiocodes and msn messenger all cause seg faults
when calling ccm thru * (or vice-versa)
~kelvin
[chan_oh323.so] => (OpenH323 Channel Driver)
== Parsing '/etc/asterisk/rtp.conf': Found
== Parsing '/etc/asterisk/oh323.conf': Found
0:00.004 OpenH323 Wrapper OpenH323 Wrapper
2006 Nov 16
2
installing asterisk for Ubuntu Synaptic
I have an Ubuntu system and went into Synaptic and checked asterisk for
installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc
and got the following output with several errors and notices. Do I need to
do more or are these ok? I expected to have some conf files in
/etc/asterisk but there is nothing there.
Thanks!
Created by Mark Spencer <markster@digium.com>
2004 Apr 27
2
Getting started woes and an archive question
Hi All,
I am a newbie and having some trouble getting Asterisk to work.
I have checked out zapata zaptel libpri and asterisk (both v1-0_stable
and regular--in separate directories). All built according to the
documentation I have found and installed correctly.
I have modified/created zaptel.conf zapata.conf as attached.
I get the following messages/error during asterisk startup.
==
2004 Dec 17
6
Realtime and PostgreSQL
Has anyone had any luck with PostgreSQL and Realtime? The realtime
instructions on voip-info seem pretty straight forward... just not woking for
me. I've included all of my config files below, and my console output.
Entire console bootup output:
[root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing
2005 Jul 18
0
Crash on reload only with autoload=no
Hi,
I've been having a little problem with my asterisk servers, I have 4
identical asterisk servers setup (same hardware, same OS, same config). Once
in a while (once or twice a day) one of the server crashes on the cron job
reload. But I realized this only happens on 3 of the 4 servers. Tried to
spot the difference between that one server that wasn't crashing. The
difference I found was
2004 Apr 02
1
error with asterisk -vvvvc
Hi
I?m a new user and I do test with my hardware
.
I have a x100p and telephone vozip.
And when I run this command asterisk ?vvvvc for to test it
.
My computer show it ?warning?
[chan_iax.so] => (Inter Asterisk eXchange)
== Manager registered action IAX1peers
== Parsing '/etc/asterisk/iax1.conf': Not found (No such file or
directory)
Apr 2 07:45:12 ERROR[16384]:
2005 Mar 29
1
Voicetronix OpenSwitch12 chan_vpb problem
Hello all,
I hope this is not off-topic, if it is please let me know.
I'm currently playing with an Asterisk at home, in order to get to know
it's ins and outs. Very very impressive indeed. I've got it hooked up to
my home phone line via a Wildcard clone board (Intel modem with Ambient
chipset), and it works like a charm. Zaptel picks up the card as a
generic clone, and works with it
2013 Jan 10
1
Segmentation fault after upgrading from asterisk-10.5.0 to asterisk-11.1.2
After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting a Segmentation fault.
[root at localhost asterisk-11.1.2]# asterisk -vvvvvvc
Asterisk 11.1.2, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components
2004 Jul 29
0
G.729 between Zap and SIP
Hi,
I have licensed the digium G.729A codec. But for some reason incoming and
outgoing calls will ALWAYS use G.711a. When I force my phone to only accept
G.729 then an incoming call from ZAP goes straight to my voicemailbox as the
phone doesn't accept the codec Asterisk wants, even if I force it in
sip.conf.
Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ?
The