Displaying 20 results from an estimated 600 matches similar to: "conference & sip"
2005 Jul 08
1
Help needed - Zap Transfer Failing...
Hi.
I have the following line in the default context of all my internal
extensions:
exten => 9876,1,Transfer(125)
When I dial extension 9876 from any sip phone, * dutifully transferrs it to
extension 125, which is just what I want.
Unfortunately when I dial 9786 from my Zap connected analogue phone, the
transfer doesn't go through and the dialplan drops through to a hangup.
debug
2004 Aug 26
4
Codec
Good day all
I want to know what the best codec is to use for asteris for VOIP
We have two towns connected with a 64k line that's going to do VOIP with
astersik.At the moment with the default installation the quality is bad and
the bandwith is high.
Is this even a codec problem
Pleas help
ALtus
2013 Nov 26
2
Unable to install Puppet Modules
Hi Puppet Users -
Currently attempting to install puppet modules on a Security Onion virtual
machine. Security Onion is based on Ubuntu 12.04.
Puppet was installed using the puppetlabs-release-precise.deb file found on
apt.puppetlabs.com.
Attempting to install a module results in the following error -
user@user-virtual-machine:~$ sudo puppet module install puppet/stdlib
--debug --verbose
2002 Nov 06
4
offline rsync
hello, I'm new to this list.
here is my question:
I would like to synchronize two computers (say the home one and the job
one) using zip drives or similar (cdroms, etc), since modem lines are
quite slow and expensive (in Italy).
I though I could produce the "signature" of files on home computer, store
it on a zip, go to job, run rsync to copy the missing or
altered files on
2005 Jul 14
5
asterisk number of calls
Good day all
What is the amount of calls that asterisk can handle,SIP and from/to
PSTN
--
Thanks
Altus Snyman
Stormcorp Network Solutions
+27 11 8071141 exten 301
2004 Sep 29
5
music on transfer
Good day all
I got my Music on hold to work but can I/how do i get music on transfer?
Please help
Thanks
2004 Sep 08
3
sendmail&hostname
Good day all
I'm just wondering for interest sake
I have a box,hostname=myname.co.za,running sendmail
If I send mail to someuser@myname.co.za it try to deliver to the box,witch
does not have the mail box.How do I tell sendmail that it should send mail to
myname.co.za's mailserver.
I know how easy it is to change the name but there's a lot of reasons why we
can.It is not in the
2004 Sep 13
5
music on hold not strting
Good day all
I added the music on hold entry in vpb.conf and commented out default line in
musiconhold.conf.
Asterisk starts up with the default mp3 but as soon as I remove it and add my
mp3 it just doenst start up and gives a broken pipe error?
Please Help or advice
Thanks
ALtus
2005 Jan 12
6
snom220
Good day all
I got my snom 220 phone so that it displays on the buttons if someone is
calling that extension
I just added "exten => 403,hint,SIP/403" in my dialplan
But
These lights only comes on if someone calls that extension,not if that
extension is busy are a call is made from that extension
Can this be done?
Please Help
Altus
2004 Aug 05
2
personal voicemail
Good day all
IS there a way to personalise the voicemail message when you leave a
message?
Thanks
Altus
2004 Aug 05
2
shared voicemail
Good day all
I got my voicemail message working,thanks but now,keep in mind I'm using
SIP
We have,for example 4 people in our admin department.Each user has its
own voicemail so that when their extension is dialed directly and not
answered it gos to voicemail.
But there is also a option to dial 3 for admin with will dial all 4
number in sequence.This I got working 100% but now I want a
2005 Sep 15
2
cdr server
Good day all
Is it possable to set asterisk up as a cdr server for other voip units
We got a quintum dx here and its got a option to log to a cdr server on
port 9002
Thanks
Altus
2005 Feb 08
2
bri dropping calls
Good day all
We have a quad bri card,installed on fedora core1,downloaded the latest
bri-stuff that download asterisk 1.0.3 and zaptel 1.0.3 and libpri 1.0.3
All installed and working.BUT
after 5min+ of talking it just drops the calls?
Any reason why?
Please help
Thanks
Altus
2004 Apr 30
2
South-Africa
Good day all
I'm in South-Africa,currently we are using openline4 cards for our pbx
systems.Now we first need approval on the cards form icasa(a government
standards) before we can use the card.The market here is very big for a
system like asterisk.The only problem is to get a card approved(for a
small company like us) its just about impossible.
Now what I'm looking for is a company that
2003 Apr 29
28
H323
Is H323 built into the current CVS? If so, could someone give me an idea of a simple config?
thanks,
darran
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2004 Nov 30
3
cisco 7902g
Can i register a cisco 7902G with asterisk?????
what i have to do???
thanks in advance.
R
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2004 Apr 05
1
sip no sound?
Good day all
So I've installed asterisk with my openline4 card and I've setup sip and
I can do sip on the local network,we are using soft clients,x-lite.
But...
When a call comes in from the outside(PSTN) and the you dial the
extension it forwards the call the the client and you see incoming call
on x-lite,you accept he call....BUT there is no sound.It shows there is
a call and you are
2004 Aug 13
2
not hangup
Good day all
I'm using sip protocol and a openline4 card.If I dial out of the pstn
and hangup a answered call it does not disconnect the connection.It
shows there is still a call on the external phone I called but on my
side its says i'm not connected.We are using x-ten soft phones
Can someone please help me
Thanks
Altus
2004 Aug 20
1
dual servers
Good day all
I'm trying to configure 2 asterisk servers running sip to connect with
each other with iax so both sip extensions can dial each other
I'm using this webpage but I'm a bit stuck
each time I try to dial the other server's sip extension it says trying
and then just gives a busy tone.In asterisk it says it could not create
aix channel and that all is busy at the moment
Can
2005 Jan 04
1
Call(out) routing
Good day all
I had a look at the extensions.conf sorting
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting
What I'm trying to do is route all my cellphone number threw a channel
and all other calls threw the other 3 channels
Cellphone numbers are 10 number,i.o.w XXXXXXXXXX.
This is what I tried but it doesn't seem to work,please help
Thanks
Altus
extensions.conf