Displaying 20 results from an estimated 6000 matches similar to: "How to share Zap channels in 2 Asterisk servers"
2004 May 22
1
Is it Possible
Hello
I am trying to setup Asterisk on 2 servers PBX300 and PBX200.
PBX300 has X100P card with 1 telephone line. PBX200 don't have any Zap device.
Softphone from PBX200 can talk to softphone on PBX300 but no out going call from PBX200.
I can call from PBX300 outside but I am unable to configure soft Phone defined in PBX200 to dial out side using PBX300 Zap devices.
I am geting error
2004 May 14
3
SoftPhone to SoftPhone with No Voice
Hello
I Installed Asterisk on RedHat 9. I am currently try to configure minimum with
two softphone talking to each other over the LAN. I am using X-Lite softphones
from xten.com site. I defined 3 phones in sip.conf and also specifies in
extensions.conf file. I am able to ring other computer but there is no voice
exchange ( i can't hear any think except ring). Here is the portion of sip.conf
2004 May 16
2
(no subject)
Hello
I am trying to configure Asterisk on RedHat Linux 9 with one X100P one port card
and one USB one port FXS card. I can modprobe wcusb but ztcfg always return
ZT_CHANCONFIG failed on channel 2: No such device or address (6)
error message.
Also unable to config outgoing call using SIP SoftPhone.
Any working examples of configuration files is highly appreciated.
I mentoned followin lines
2005 May 26
1
Using zap channels on 2 different servers
Let say I have a server located in Europe and one in North America.
The 2 servers are connected together with iax2.
Both server are connected to phone lines in there own country.
If I want that when a user call a north american phone number from
the server in Europe it use a zap channel on the server located in
North America and also if someone in North America dial an European
phone
2007 Feb 13
16
Error against latest trunk while testing via spec for model
Hi
I just did an update to lates trunk
=================
context "Given a generated venue_spec.rb with fixtures loaded" do
fixtures :venues
specify "fixtures should load two Venues" do
Venue.should have(2).records
end
end
==================
gives me
==========
1)
TypeError in ''Given a generated venue_spec.rb with fixtures loaded
fixtures should load two
2009 Dec 02
4
http url support in swfdec-0.8.4
Dear All,
I am using swfdec-0.8.4 player over directfb.. I am able to play stored swf
files where as im not able to play any only swf files
I get following error
open http://www.foo.com/foobar.swf failure, err = No such file or directory
Thanks and regards
Deepak
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2012 Oct 05
4
PuppetDB Installation
I am attempting to install PuppetDB from source.
I am a bit confused however, is lein required?
thanks!
--
You received this message because you are subscribed to the Google Groups "Puppet Users" group.
To post to this group, send email to puppet-users@googlegroups.com.
To unsubscribe from this group, send email to puppet-users+unsubscribe@googlegroups.com.
For more options, visit
2007 Jan 30
5
errors while testing resource controller using rpec
I am testing a resource called venue in this piece of code (generated
using script/rspec_resource)
====================
context "Requesting /venues using POST" do
controller_name :venues
setup do
@mock_venue = mock(''Venue'')
@mock_venue.stub!(:save).and_return(true)
@mock_venue.stub!(:to_param).and_return(1)
Venue.stub!(:new).and_return(@mock_venue)
2005 Jul 27
2
oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6
Abwesenheitsnotiz: [Asterisk-Users] oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6Hi All
I am using oh323 with 6.6 virsion , and runing under xeon 2.8 dual with 2 gb
ram, with g729 for i686 , (fedora 1).
my problem is sip - oh323 - h323 (quintum) - pstn , sip party can listen
otherparty realtime voice , but other party geting sip party's voice 1 sec
later (not
2007 Oct 29
12
Reports in Puppet
Hi,
I wanted to configure reporting in puppet. I came across the
below links which says what needs to be added in config file to enable
reports.
But I was wondering how I view the visual report/graphs do I need to
create my own script.
http://reductivelabs.com/trac/puppet/wiki/ReportReference#rrdgraph
http://reductivelabs.com/trac/puppet/wiki/ReportsAndReporting
--
Deepak
2004 Sep 30
2
Win2003 ADS member server - almost working, ideas?
I am attempting to install a Samba-3.0.0,1 on FreeBSD
5.2.1-RELEASE server to an existing Windows 2003
Server Active Directory Domain.
I've followed Chapter 6 of the HOWTO man to get as far
as I have.
#kinit gooduser --successfully gets a kerberos
ticket
#wbinfo --authenticate=gooduser%goodpassword --
successfully authenticates all user accounts (that
I've tested)
#wbinfo -u yields
2007 Mar 31
3
Help in getting aggregated data
Hi team,
I have the data of the form:
> a<- data.frame(x=c(1,2,1,4,3), y=c(1,2,1,4,3), z=c(1,2,3,4,5))
I need the output of the form
> b<- data.frame(x=c(1,2,3,4), y=c(1,2,3,4), z=(3,2,5,4) )
As you can see, the Z value contains the maximum for each of the (x,y)
combinations.
I used
> c<-by(a$z, list(x=a$x, y=a$y), max)
> c[,]
y
x 1 2 3 4
1 3 NA NA NA
2
2007 Jun 08
11
Bad Echo between SIP calls
Hi,
We have a PRI connection & when its was on test networks we had echo problems withoutside line.
So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 & RHEL4-Update 4.
But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear.
I am not sure whats
2012 Feb 29
2
peer probe fails
Hi,
Unable to do peer probe... and unable to figure out whats the
reason from the gluster log.
can someone help ?
1) This is what i was trying...
gluster> peer probe llm19.in.ibm.com
Probe unsuccessful
Probe returned with unknown errno 107
gluster> peer probe 9.124.111.25
Probe unsuccessful
Probe returned with unknown errno 107
gluster> peer status
Number of Peers: 1
Hostname:
2012 Mar 09
3
Interacting with the Operating System
Is there any way to issue operating system commands and geting back the results,
in R?
I mean, for instance, in Linux, to execute from R the 'ls' command and getting
back a list of files in the current directory, or, equivalently, in Windows/DOS,
the 'dir' command?
I'm not interested in the 'ls' or 'dir' commands it is just an example.
Do you have any
2013 May 07
7
puppet node clean using SQLite instead of PuppetDB
I have a node that has some bad stored configs (namely the wrong ssh host keys) that I''m trying to clear out. Looking around it seems I''m supposed to do:
puppet node clean foo.example.com
However, that keeps bailing out because it''s trying to open a SQLite3 db where stored configs are normally kept, but my stored configs are kept in puppetdb (http://pastie.org/7814483
2007 Oct 02
6
Push /home/* directories recursively to clients
Hi I am trying to push populate /home & subdirectories from the puppet
server to all the Linux clients.
I managed this with cfengine using rsync. But I am not sure how do I
achieve this in puppet, do we have any inbuilt function for this.
Also, is there a function for userdel like for useradd (user)
groupadd(group).
Any suggestion is appreciated.
--
Deepak
2007 Oct 19
3
Puppet port/install on FreeBSD & Open BS
Does anyone know how to install puppet as client on FreeBSD & OpenBSD.
--
Deepak
_______________________________________________
Puppet-users mailing list
Puppet-users@madstop.com
https://mail.madstop.com/mailman/listinfo/puppet-users
2007 Sep 13
2
Paging to external speaker like in airports etc...
Hi, I have a production asterisk-1.2.8 system with FreePBX & PRI Digium card.
I am looking for a paging system to an external speaker. I can page to internal Polycom 501 VoIP.
But, what hardware or system do I need to integrate with the asterisk to have this acheived.
--
Deepak
Linux your Life, Don't Window it [[]]
{ All for the best }
2014 Sep 11
1
chan_sip.c:23647 handle_request_invite: Failed to authenticate device
Hi,
Why are we getting message in the asterisk
[Sep 10 12:55:23] NOTICE[15043]: chan_sip.c:23647 handle_request_invite:
Failed to authenticate device 601<sip:601 at 111.118.185.107>;
tag=2f498fbd
[Sep 10 12:55:24] NOTICE[15043]: chan_sip.c:23647 handle_request_invite:
Failed to authenticate device 601<sip:601 at 111.118.185.107>;tag=209a8aa9
Regards
Deepak Bhatia
--------------