Displaying 20 results from an estimated 10000 matches similar to: "Minimal hardware requirements"
2005 Feb 14
18
Which IP phone to use in Australia
Hi, all
I am in Australia and I have to setup Asterisk in few offices. There will be IP phones in each office and I must be able to call between offices.
I need actual handsets. I need "standard" handsets to be used by people. Those must support features like CID, call forward, etc. --- your normal office feature set.
Also I need some sort of more complex handset to be used by
2003 Apr 03
5
Hardware requirements
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2007 Jan 23
12
How to exit from console?
Hi, all
Stupid question, but how do you exit asterisk console without stopping
the asterisk?
Tried quit and exit:
*CLI> exit
No such command 'exit' (type 'help' for help)
*CLI> quit
No such command 'quit' (type 'help' for help)
*CLI>
Any other ideas?
I started asterisk with -cvvvvg option. Same problem if use asterisk
-r to connect. Can not exit.
Any
2005 Jul 14
4
Polycom configs?
I have a number of Polycom phones to setup with my * server. For my
initial few phones I hand wrote configs. Does anyone here who uses
Polycom phones have some form of management utility for automating
their setup?
Michael
--
Michael Graves mgraves@pixelpower.com
Sr. Product Specialist www.pixelpower.com
Pixel Power Inc.
2006 Dec 14
4
Zaptel under FC6
Hi, all
I am building a new server. Have installed FC 6 and put in TDM400 card.
Checked out latest asteriusk code, run make install in zaptel directory.
So far all is fine.
Now I am trying to install the drivers.
# modprobe zaptel
FATAL: Module zaptel not found.
Fair enough, no zaptel driver is found on the system.
Is there are any known problems with FC6? I did not have much trouble
running
2005 Sep 10
2
VoipBuster again
Hi, all
I am still battling to connect * and voipbuster.
What protocol does it use? Ethereal capture shows UDP traffic, but no SIP or
IAX traffic when using their client.
VoipBuster client connects to connectionserver.voipbuster.com on port 11112
for authentication. Call itself is placed on different server.
I have tried to connect using SIP and IAX and it seems that no
authentication is
2005 Aug 12
3
OT: Sendmail question
Hi, all
I want voicemails to be delivered to recepients by e-mail. I have set up
voicenail, and I can see asterisk is using sendmail to send messages out.
Using Ethereal, I can see that messages are leaving my network, but
receipeint mail server never replies back. As a result, mail delivery is
timed out.
I got a book on sendmail and it looks quite complex. It will take quite a
bit of time
2005 Aug 13
14
Why NAT problem
hello
i am using asterisk-1.0.9. i have a NAT problem.
without NAT registration is ok. and if user is bhind
NAT it is registring on asterisk. but SJPhone is
showing "not registered". i think asterisk is properly
sending request to UA. any comments............this
sip.conf setting was working previously
-- Registered SIP '5000' at 0.0.0.0 port 5060
expires 120
-- Saved
2003 Jul 11
8
G729 codec problems
Hi I seem to have installed the G729 codec properly
but can't seem to get it to work ..
in the Asterisk startup I see ..
[codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator)
== Detected 1 licensed G.729 transcoders
WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator
'g729tolinb' does not produce sample frames.
== Registered translator
2008 Jan 25
2
SPA3000 -- PSTN to VoIP
Hi, all
I am trying to figure out how to forward incoming PSTN call on SPA3000
to VoIP extension(s).
Basically, I have converted my home to VoIP. I have normal phone
(connected to SPA3000) and couple of IP phones. All call coming from
VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I
need to do same thing for incoming PSTN calls.
I have enabled gateway function in SPA3000 and
2005 Jul 16
2
beginners question about extension context
Hi, all
I have couple of SIP phones and they are in [from-sip] context.
I also have an IAX2 phone. I have put this one in [iax-user] context.
I want to make calls between SIP and IAX2 phones. If I put them all in same
context all is fine, however when they are in different contexts they will
not call each other and I will get message (in * CLI) that particular
extension does not exist in a
2005 Feb 22
13
TFTP Server
G'Day All,
Can anyone give me some direction in setting up the TFTP server on my
RadHat ES3 box?
I did quite a bit of reading, but I think I am more unsure now than
before. I found the information nebulous. TFTP is already installed. I
am trying to determine where the root directory for the tftp services is
located so I can copy the CISCO 7960 firmware files onto it.
Thanks.... Ferg
2005 Oct 10
2
TDM400 not working
Hi, all
I have installed TDM400 card. I can see it is there (lspci).
But Asterisk does not find it.
phonebox2*CLI> zap show status
No Zaptel interface found.
I assume that driver is not loaded, but I am sure I have installed it (I
compiled zaptel and then re-build asterisk and did make install for both
zaptel and asterisk).
When asterisk is started I get:
Jan 2 06:28:08 WARNING[3473]:
2005 May 12
5
chan_capi, chan_misdn and chan_modem
Could someone please comment on the current state of chan_capi,
chan_misdn and chan_modem channel drivers in terms of functionality and
stability. Specifically, which channel driver would be best for a
passive PCI HFC or W6692 ISDN card. The chan_misdn wiki claims that
chan_capi distinguishes between junghanns and non-junghans cards, and
that chan_misdn is better suited for general misdn
2005 May 10
13
What do you name yours
Hello list
we are installing 2 new servers (to run asterisk) shortly, for a
"stand alone" service. Ignoring our current naming convention, we'd
like to name them something.. but we are not sure what.
a consideration is that on the screens of the phones it shows
extension@hostname (eg 3001@telephony) (all extensions are numeric) so
the users will see it everyday
i'm not
2005 Sep 18
2
HW Question (TDM400)
Hi, all
Did anyone tried TDM400 card on old main board (Intel 440LX chipset -- PII)? The reason I am asking is because TDM400 needs PCI2.2 and main board is PCI2.1
I do not want to upgrade yet.
Thanks,
Rudolf
2005 Aug 06
1
Voicemail -- newbie question
Hi, all
I am trying to set up voicemail. I've done it to the point where I can leave
messages.
How do I retrieve them?
Actually I have few questions:
1. I want voice mail to be available at certain extension, say 100. How do I
set it up so all users can call this number and get to their respective
mailboxes.
2. How do I let users to create their own voicemail passwords from the
phone?
3.
2005 Feb 14
7
Outbound Caller ID on PRI
We have just had an ISDN PRI service commissioned here in AU (using Powertel
as provider). I have called them and ensured that we have the ability to set
Caller ID on our service to any number in our 100 number block, and I have
been assured that everything is OK from their end (unlikely). Every time I
drop a sample.call in to /var/spool/asterisk/outgoing with the Callerid:
option set, the
2005 Oct 08
2
Configuring TDM400 in Australia
Hi, all
I have installed TDM400 with 1 FXS and 1 FXP ports.
Now I am goig through documentation on how to configure it.
It mentions 3 protocols: Loopstart, Groundstart and Koolstart. Which one do
I use?
Can someone send me sample zaptel.conf file for Australia? This will save me
some time and will be used as a working example.
Thanks,
Rudolf
2004 Sep 22
1
Opteron vs Xeon?
Hi
I'm setting up a SIP gateway, serving quite a few potential users, and
I wonder if I should purchase a Opteron or Xeon based system. Xeon has
it's HT, but is it worth it? Has anyone tested Opteron on asterisk?
Does it work well?
There'll be no transcoding in this system - G.711A all way through
roy