Displaying 20 results from an estimated 6000 matches similar to: "Permission denied when obtaining Status"
2004 Jul 13
2
IAX2 calls through IAXTEL.com
I created an account at IAXTEL.com to route 1-700-XXX-XXXX calls
through. IAXTEL.com gave me a number (example) of 700-555-6226. I have
made the following changes to my:
/etc/asterisk/extensions.conf:
[iaxtel700]
exten =>
_81700XXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1})
exten =>
_81800NXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1})
2007 Mar 07
0
gtalk2voip and Asteris
What kinds of problems were you having? I'm on 1.4.0 and chan_gtalk.so
simply doesn't load. Of the 146 files in the /usr/lib/asterisk/modules/
directory, asterisk loads 144 of them, omitting only chan_gtalk.so and
res_jabber.so.
Connected to Asterisk 1.4.1 currently running on monkey (pid = 9371)
Verbosity is at least 3
foo*CLI> module load chan_gtalk.so
[Mar 7 10:23:07]
2006 Jan 19
0
Incoming fax on voipbuster
Hello,
I'm trying to receive a fax to my inbound number from voipbuster.
Asterisk receives the call and starts the rxfax application successful,
but then nothing happens. The calling party is still hearing a ringing
tone, or sometimes nothing. Voicecalls are working correct and without
problems.
For testing I've add a local number (300) to the dialplan. When I call
this number
2010 Jun 10
1
Am I having problems with codecs? or am I not receiving an invite at all from my DID provider?
Hi Guys,
I have Spikko setup as provider of DID and outbound routes and I can make
calls out but no inbound calls via DID can be made. I did a sip debug which
is reported below. I never receive the call though, I have a catch all in my
inbound routes and it doesn't hit my context at all or not sip invite comes
in:
FreePBX:
Trunk Name:
*Spikko*
Peer Detail
*username=MyUsername*
2004 Jun 23
5
Really basic stuff :(
Hi :)
I've had all this working before, but I'm revisiting it, and in short, I
currently have huge problems receiving incoming calls. I've been trying
with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel
and libpri as of yesterday afternoon.
Would someone mind helping? :)
My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set
as the 'DMZ
2003 Jul 07
0
Follow-up -- Using Asterisk with Nikotel
Hi
thanks to everybody who has been assisting me in solving the various
problems I had to dial out from Asterisk to a PSTN number with SIP using
Nikotel's VoIP service.
I have drafted a mini-how-to which is available at
http://www.akabeni.com/benjk/Using_Asterisk_with_Nikotel.pdf
This is a first draft, I will amend this further, in particular the
"verify and debug" section
2003 Apr 30
0
XP Home Domain Login Workaround
Hello, I administer a small office. As is common, we have samba acting as
Domain controller and I have a 'logon script' configured in smb.conf.
The logon script does not do much: map the P: (public) and H: (home) drives, as
well as setting the time. This is done through net commands, and is all very
familiar to readers of this list.
The new laptop arrived from Dell, and it's XP Home.
2008 Dec 02
2
Samba ADS Error "Session setup failed: Call returned zero bytes (EOF)"
Hi Samba Bods,
Sorry for re-posting this one but I got no response to my last post except
for a level 10 logs request which I uploaded last week.
I have been looking at numerous howtos and newsgroup postings and I cannot
spot what the issue is. I am sure its a simple config issue, but I am lost
..
I am using Samba 3.2.4 compiled from source on AIX 5.3 TL8 and using
"security =
2006 Jan 26
0
XML Request in R: Pointers/examples needed
Hello R-helpers,
I'm new to XML. I have been using an application for some time, but now I
wish to automate my downloads with R. When I use the web interface of their
XML application, I'm able to read the response in R with the XML package.
My problem now is to send the requests directly from R. Here below are two
XML request scripts that come from the applicaton help file. My
2008 Nov 26
3
AIX 53TL8 Samba 3.2.4 Active Directory Win2k3 - "session setup failed: Call returned zero bytes (EOF)"
Hi All,
I am using Samba 3.2.4 compiled from source on AIX 5.3 TL8 and using
"security = SERVER" in the smb.conf works fine, however I am having
some issues when using "security = ADS" ..
I have followed numerous HOWTOs and newsgroup listings and seem to be
going round in circles ..
I think I can authenticate ok against the domain win2k3 server, but
then Samba bombs out with
2011 Jul 26
0
RADIUS Questions
I've been running FreeRadius 2 on Centos 5.5 for a while now. So far so
good. I'm now looking to make connecting to our WPA secured wireless easier.
The RADIUS server is running in a VM and since the system is in use I
have copied the original and used that copy to create a test
environment. I have run through all system updates and have upgraded all
relevant packages. The test system
2013 Mar 12
1
How does Asterisk handle ACK's?
Hello,
I'm noticing strange behavior in one of our Asterisk nodes where the ACK is always sent to the proxy, but RR is not enabled for calls.
The proxy drops the ACK.
I'm using the AMI interface to originate a call:
Action: login
Username: myusername
Secret: mypassword
Events: on
Action: Originate
Channel: SIP/<SOMENUMBER>@proxy1
CallerID: <SOMENUMBER>
Application: Playback
2004 Jun 01
1
SIP vs. SIP :-(
I'v a sip client and a sip trunk to FWD:
[general]
port=5060
context=default
tos=reliability
disallow=all
allow=ulaw
careinvite=no
[freeworlddialup]
context=default
type=friend
username=MYUSERNAME
secret=MYPASSWORD
host=fwd.pulver.com
[igor]
type=friend
callerid="Me"
host=dynamic
dtmfmode=rfc2833
careinvite=no
When i try to call a FWD number from SIP client i obtain a lot of
2019 Apr 04
2
Message: Authentication failed on manager interface
I'm not sure how much more simple I can make this but I just cannot
seem to get my Asterisk 13 to accept a connection on the manager
interface:
--- manager.conf ---
[general]
enabled = yes
port = 5038
bindaddr = 127.0.0.1
[myasterisk]
secret=a
permit=0.0.0.0/0.0.0.0
read = all
write = all
So, couldn't be any more wide open and simpler to connect yet:
# echo -e "Action:
2005 Jul 06
1
/etc/asterisk/manager.conf
Valued Colleagues,
I am trying to configure and use asterisk manager API.
The /etc/asterisk/manager.conf and the output of "netstat -nl" are
appended below.
When I restart asterisk, I believe I should be able to see the asterisk
listening on
port 5038 using netstat. But when I type netstat, I don't see any
applications listening
on port 5038.
When I telnet to port
2005 Apr 05
1
Can't mount samba share, Access denied
Hello,
I have samba configured with the following smb.conf file:
[global]
workgroup = mydomain
netbios name = servername
security = domain
printcap name = cups
disable spoolss = yes
show add printer wizard = no
idmap uid = 15000-20000
idmap gid = 15000-20000
winbind use default domain = yes
use sendfile = yes
printing = cups
[myshare]
comment = My new share
path = /export/myshare
valid users =
2002 Mar 14
4
Samba client issues
Hello everyone,
I am a newbie to samba. I am trying to mount a Windows 2000 share directory on my linux box using smbmount. I am able to do so successfully. But I do see some errors pop up when I run the command. My mount seems to be stable and it is mounted as rw.
# smbmount "//EngineerWKS85/My Drivers" /mnt/samba -o username=MyUsername,password=MyPassword,workgroup=MyWorkgroup
2010 Oct 25
4
google voice + asterisk: calls made to GV# processed but weird
Dear all,
First off, I am very new to asterisk so forgive me if any of my
comments or questions seem trivial. Thanks to [this
post](http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/)
and [this post](http://www.davidvossel.com/?p=28), I have GV set up on
asterisk through jabber.conf and gtalk.conf. I can successfully dial
out from asterisk.
I'm trying to set up an
2004 Jan 29
1
re: help with voicepulse connect IAX2
hello,
after playing with an asterisk configuration for voip for a few weeks i'm
trying to get outbound dialing with voicepulse going - i've cut down the
asterisk to a very minimal install (1 SIP client) to try to localize the
problem. The SIP client works fine (SIP and * on the same NAT) and could
access the demo from samples before i removed it, and can call itself - so
i am
2014 Aug 18
2
AMI & Elastix
Hi all!
I have trouble with connection to AMI 1.1 wich enabled on Elastix
"*Asterisk Call Manager/1.1*
*Action: Login Username: admin Secret: qweasd123*
*Response: Error*
*Message: Missing action in request*"
Elastix versions:
"* Kernel*
* Linux(x86_64)-2.6.18-348.1.1.el5*
* Elastix*
* elastix-2.4.0-1*
* elastix-portknock-0.0.1-0*
* elastix-agenda-2.4.0-1*
*